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webrtc/talk/session/media/mediasessionclient_unittest.cc

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Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 02:45:36 +02:00
/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include <string>
#include <vector>
#include "talk/base/gunit.h"
#include "talk/base/logging.h"
#include "talk/base/scoped_ptr.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/devices/fakedevicemanager.h"
#include "talk/p2p/base/constants.h"
#include "talk/p2p/client/basicportallocator.h"
#include "talk/session/media/mediasessionclient.h"
#include "talk/xmllite/xmlbuilder.h"
#include "talk/xmllite/xmlelement.h"
#include "talk/xmllite/xmlprinter.h"
#include "talk/xmpp/constants.h"
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-10 09:18:04 +02:00
using cricket::AudioCodec;
using cricket::AudioContentDescription;
using cricket::Codec;
using cricket::DataCodec;
using cricket::DataContentDescription;
using cricket::FeedbackParam;
using cricket::FeedbackParams;
using cricket::VideoCodec;
using cricket::VideoContentDescription;
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 02:45:36 +02:00
// The codecs that our FakeMediaEngine will support. Order is important, since
// the tests check that our messages have codecs in the correct order.
static const cricket::AudioCodec kAudioCodecs[] = {
cricket::AudioCodec(103, "ISAC", 16000, -1, 1, 18),
cricket::AudioCodec(104, "ISAC", 32000, -1, 1, 17),
cricket::AudioCodec(119, "ISACLC", 16000, 40000, 1, 16),
cricket::AudioCodec(99, "speex", 16000, 22000, 1, 15),
cricket::AudioCodec(97, "IPCMWB", 16000, 80000, 1, 14),
cricket::AudioCodec(9, "G722", 16000, 64000, 1, 13),
cricket::AudioCodec(102, "iLBC", 8000, 13300, 1, 12),
cricket::AudioCodec(98, "speex", 8000, 11000, 1, 11),
cricket::AudioCodec(3, "GSM", 8000, 13000, 1, 10),
cricket::AudioCodec(100, "EG711U", 8000, 64000, 1, 9),
cricket::AudioCodec(101, "EG711A", 8000, 64000, 1, 8),
cricket::AudioCodec(0, "PCMU", 8000, 64000, 1, 7),
cricket::AudioCodec(8, "PCMA", 8000, 64000, 1, 6),
cricket::AudioCodec(126, "CN", 32000, 0, 1, 5),
cricket::AudioCodec(105, "CN", 16000, 0, 1, 4),
cricket::AudioCodec(13, "CN", 8000, 0, 1, 3),
cricket::AudioCodec(117, "red", 8000, 0, 1, 2),
cricket::AudioCodec(106, "telephone-event", 8000, 0, 1, 1)
};
static const cricket::VideoCodec kVideoCodecs[] = {
cricket::VideoCodec(96, "H264-SVC", 320, 200, 30, 1)
};
static const cricket::DataCodec kDataCodecs[] = {
cricket::DataCodec(127, "google-data", 0)
};
const std::string kGingleCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1'> " \
" <usage/> " \
" <rtp:crypto tag='145' crypto-suite='AES_CM_128_HMAC_SHA1_32'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='AES_CM_128_HMAC_SHA1_80'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
// Jingle offer does not have any <usage> element.
const std::string kJingleCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1'> " \
" <rtp:crypto tag='145' crypto-suite='AES_CM_128_HMAC_SHA1_32'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='AES_CM_128_HMAC_SHA1_80'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
const std::string kGingleRequiredCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1' required='true'> "\
" <usage/> " \
" <rtp:crypto tag='145' crypto-suite='AES_CM_128_HMAC_SHA1_32'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='AES_CM_128_HMAC_SHA1_80'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
const std::string kJingleRequiredCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1' required='true'> "\
" <rtp:crypto tag='145' crypto-suite='AES_CM_128_HMAC_SHA1_32'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='AES_CM_128_HMAC_SHA1_80'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
const std::string kGingleUnsupportedCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1'> " \
" <usage/> " \
" <rtp:crypto tag='145' crypto-suite='NOT_SUPPORTED_1'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='NOT_SUPPORTED_2'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
const std::string kJingleUnsupportedCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1'> " \
" <rtp:crypto tag='145' crypto-suite='NOT_SUPPORTED_1'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='NOT_SUPPORTED_2'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
// With unsupported but with required="true"
const std::string kGingleRequiredUnsupportedCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1' required='true'>" \
" <usage/> " \
" <rtp:crypto tag='145' crypto-suite='NOT_SUPPORTED_1'" \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/>" \
" <rtp:crypto tag='51' crypto-suite='NOT_SUPPORTED_2'" \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
const std::string kJingleRequiredUnsupportedCryptoOffer = \
"<rtp:encryption xmlns:rtp='urn:xmpp:jingle:apps:rtp:1' required='true'>" \
" <rtp:crypto tag='145' crypto-suite='NOT_SUPPORTED_1' " \
" key-params='inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9'/> " \
" <rtp:crypto tag='51' crypto-suite='NOT_SUPPORTED_2' " \
" key-params='inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy'/>" \
"</rtp:encryption> ";
const std::string kGingleInitiate(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='104' name='ISAC' clockrate='32000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='119' name='ISACLC' clockrate='16000' bitrate='40000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='99' name='speex' clockrate='16000' bitrate='22000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='97' name='IPCMWB' clockrate='16000' bitrate='80000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='9' name='G722' clockrate='16000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='102' name='iLBC' clockrate='8000' bitrate='13300' />" \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='98' name='speex' clockrate='8000' bitrate='11000' />" \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='3' name='GSM' clockrate='8000' bitrate='13000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='100' name='EG711U' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='101' name='EG711A' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='0' name='PCMU' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='8' name='PCMA' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='126' name='CN' clockrate='32000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='105' name='CN' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='13' name='CN' clockrate='8000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='117' name='red' clockrate='8000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='106' name='telephone-event' clockrate='8000' /> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiate(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" <payload-type id='104' name='ISAC' clockrate='32000'/> " \
" <payload-type " \
" id='119' name='ISACLC' clockrate='16000'> " \
" <parameter name='bitrate' value='40000'/> " \
" </payload-type> " \
" <payload-type " \
" id='99' name='speex' clockrate='16000'> " \
" <parameter name='bitrate' value='22000'/> " \
" </payload-type> " \
" <payload-type " \
" id='97' name='IPCMWB' clockrate='16000'> " \
" <parameter name='bitrate' value='80000'/> " \
" </payload-type> " \
" <payload-type " \
" id='9' name='G722' clockrate='16000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='102' name='iLBC' clockrate='8000'> " \
" <parameter name='bitrate' value='13300'/> " \
" </payload-type> " \
" <payload-type " \
" id='98' name='speex' clockrate='8000'> " \
" <parameter name='bitrate' value='11000'/> " \
" </payload-type> " \
" <payload-type " \
" id='3' name='GSM' clockrate='8000'> " \
" <parameter name='bitrate' value='13000'/> " \
" </payload-type> " \
" <payload-type " \
" id='100' name='EG711U' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='101' name='EG711A' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='0' name='PCMU' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='8' name='PCMA' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='126' name='CN' clockrate='32000' /> " \
" <payload-type " \
" id='105' name='CN' clockrate='16000' /> " \
" <payload-type " \
" id='13' name='CN' clockrate='8000' /> " \
" <payload-type " \
" id='117' name='red' clockrate='8000' /> " \
" <payload-type " \
" id='106' name='telephone-event' clockrate='8000' /> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate string with a different order of supported codecs.
// Should accept the supported ones, but with our desired order.
const std::string kGingleInitiateDifferentPreference(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='104' name='ISAC' clockrate='32000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='97' name='IPCMWB' clockrate='16000' bitrate='80000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='9' name='G722' clockrate='16000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='119' name='ISACLC' clockrate='16000' bitrate='40000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='99' name='speex' clockrate='16000' bitrate='22000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='100' name='EG711U' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='101' name='EG711A' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='0' name='PCMU' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='8' name='PCMA' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='102' name='iLBC' clockrate='8000' bitrate='13300' />" \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='3' name='GSM' clockrate='8000' bitrate='13000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='98' name='speex' clockrate='8000' bitrate='11000' />" \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='126' name='CN' clockrate='32000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='105' name='CN' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='13' name='CN' clockrate='8000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='117' name='red' clockrate='8000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='106' name='telephone-event' clockrate='8000' /> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateDifferentPreference(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='104' name='ISAC' clockrate='32000'/> " \
" <payload-type " \
" id='97' name='IPCMWB' clockrate='16000'> " \
" <parameter name='bitrate' value='80000'/> " \
" </payload-type> " \
" <payload-type " \
" id='9' name='G722' clockrate='16000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='119' name='ISACLC' clockrate='16000'> " \
" <parameter name='bitrate' value='40000'/> " \
" </payload-type> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" <payload-type " \
" id='99' name='speex' clockrate='16000'> " \
" <parameter name='bitrate' value='22000'/> " \
" </payload-type> " \
" <payload-type " \
" id='100' name='EG711U' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='101' name='EG711A' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='0' name='PCMU' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='8' name='PCMA' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
" id='102' name='iLBC' clockrate='8000'> " \
" <parameter name='bitrate' value='13300'/> " \
" </payload-type> " \
" <payload-type " \
" id='3' name='GSM' clockrate='8000'> " \
" <parameter name='bitrate' value='13000'/> " \
" </payload-type> " \
" <payload-type " \
" id='98' name='speex' clockrate='8000'> " \
" <parameter name='bitrate' value='11000'/> " \
" </payload-type> " \
" <payload-type " \
" id='126' name='CN' clockrate='32000' /> " \
" <payload-type " \
" id='105' name='CN' clockrate='16000' /> " \
" <payload-type " \
" id='13' name='CN' clockrate='8000' /> " \
" <payload-type " \
" id='117' name='red' clockrate='8000' /> " \
" <payload-type " \
" id='106' name='telephone-event' clockrate='8000' /> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-10 09:18:04 +02:00
const std::string kJingleInitiateWithRtcpFb(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'> " \
" <rtcp-fb type='nack'/> " \
" </payload-type> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='video'> " \
" <rtcp-fb type='nack'/> " \
" <payload-type id='99' name='H264-SVC'> " \
" <rtcp-fb type='ccm' subtype='fir'/> " \
" <parameter name='height' value='200'/> " \
" <parameter name='width' value='320'/> " \
" <parameter name='framerate' value='30'/> " \
" </payload-type> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test data'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='data'> " \
" <rtcp-fb type='nack'/> " \
" <payload-type id='127' name='google-data'> " \
" </payload-type> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 02:45:36 +02:00
const std::string kGingleVideoInitiate(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/video'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/video' " \
" id='99' name='H264-SVC' framerate='30' " \
" height='200' width='320'/> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleVideoInitiate(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='video'> " \
" <payload-type id='99' name='H264-SVC'> " \
" <parameter name='height' value='200'/> " \
" <parameter name='width' value='320'/> " \
" <parameter name='framerate' value='30'/> " \
" </payload-type> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kJingleVideoInitiateWithRtpData(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='video'> " \
" <payload-type id='99' name='H264-SVC'> " \
" <parameter name='height' value='200'/> " \
" <parameter name='width' value='320'/> " \
" <parameter name='framerate' value='30'/> " \
" </payload-type> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test data'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='data'> " \
" <payload-type id='127' name='google-data'/> " \
" <rtcp-mux/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kJingleVideoInitiateWithSctpData(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='video'> " \
" <payload-type id='99' name='H264-SVC'> " \
" <parameter name='height' value='200'/> " \
" <parameter name='width' value='320'/> " \
" <parameter name='framerate' value='30'/> " \
" </payload-type> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test data'> " \
" <description xmlns='google:jingle:sctp' media='data'> " \
" <stream sid='1'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kJingleVideoInitiateWithBandwidth(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='video'> " \
" <payload-type id='99' name='H264-SVC'> " \
" <parameter name='height' value='200'/> " \
" <parameter name='width' value='320'/> " \
" <parameter name='framerate' value='30'/> " \
" </payload-type> " \
" <bandwidth type='AS'>42</bandwidth> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kJingleVideoInitiateWithRtcpMux(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" <content name='test video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='video'> " \
" <payload-type id='99' name='H264-SVC'> " \
" <parameter name='height' value='200'/> " \
" <parameter name='width' value='320'/> " \
" <parameter name='framerate' value='30'/> " \
" </payload-type> " \
" <rtcp-mux/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate string with a combination of supported and unsupported codecs
// Should accept the supported ones
const std::string kGingleInitiateSomeUnsupported(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='97' name='ASDFDS' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='102' name='1010' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='107' name='DFAS' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='100' name='EG711U' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='101' name='EG711A' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='0' name='PCMU' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='110' name=':)' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='13' name='CN' /> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateSomeUnsupported(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'> " \
" <payload-type " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type " \
" id='97' name='ASDFDS' /> " \
" <payload-type " \
" id='102' name='1010' /> " \
" <payload-type " \
" id='107' name='DFAS' /> " \
" <payload-type " \
" id='100' name='EG711U' /> " \
" <payload-type " \
" id='101' name='EG711A' /> " \
" <payload-type " \
" id='0' name='PCMU' /> " \
" <payload-type " \
" id='110' name=':)' /> " \
" <payload-type " \
" id='13' name='CN' /> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kGingleVideoInitiateWithBandwidth(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/video'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/video' " \
" id='99' name='H264-SVC' framerate='30' " \
" height='200' width='320'/> " \
" <bandwidth type='AS'>42</bandwidth> " \
" </description> " \
" </session> " \
"</iq> ");
// Initiate string without any supported codecs. Should send a reject.
const std::string kGingleInitiateNoSupportedAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='123' name='Supercodec6000' /> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateNoSupportedAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type " \
" id='123' name='Supercodec6000' /> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate string without any codecs. Assumes ancient version of Cricket
// and tries a session with ISAC and PCMU
const std::string kGingleInitiateNoAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateNoAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// The codecs are supported, but not at the given clockrates. Should send
// a reject.
const std::string kGingleInitiateWrongClockrates(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='8000'/> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='97' name='IPCMWB' clockrate='1337'/> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='102' name='iLBC' clockrate='1982' /> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateWrongClockrates(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type " \
" id='103' name='ISAC' clockrate='8000'/> " \
" <payload-type " \
" id='97' name='IPCMWB' clockrate='1337'/> " \
" <payload-type " \
" id='102' name='iLBC' clockrate='1982' /> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// The codecs are supported, but not with the given number of channels.
// Should send a reject.
const std::string kGingleInitiateWrongChannels(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' channels='2'/> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='97' name='IPCMWB' channels='3'/> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateWrongChannels(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type " \
" id='103' name='ISAC' channels='2'/> " \
" <payload-type " \
" id='97' name='IPCMWB' channels='3'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate with a dynamic codec not using webrtc default payload id. Should
// accept with provided payload id.
const std::string kGingleInitiateDynamicAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='123' name='speex' clockrate='16000'/> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateDynamicAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type " \
" id='123' name='speex' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate string with nothing but static codec id's. Should accept.
const std::string kGingleInitiateStaticAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='3' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='0' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='8' /> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateStaticAudioCodecs(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type id='3' /> " \
" <payload-type id='0' /> " \
" <payload-type id='8' /> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate with payload type-less codecs. Should reject.
const std::string kGingleInitiateNoPayloadTypes(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" name='ISAC' clockrate='16000'/> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateNoPayloadTypes(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate'> " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type name='ISAC' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
// Initiate with unnamed dynamic codces. Should reject.
const std::string kGingleInitiateDynamicWithoutNames(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <session xmlns='http://www.google.com/session' type='initiate'" \
" id='abcdef' initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/phone'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='100' clockrate='16000'/> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleInitiateDynamicWithoutNames(
"<iq xmlns='jabber:client' from='me@domain.com/resource' " \
" to='user@domain.com/resource' type='set' id='123'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-initiate'> " \
" sid='abcdef' initiator='me@domain.com/resource'> " \
" <content name='test audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'>" \
" <payload-type id='100' clockrate='16000'/> " \
" </description> " \
" <transport xmlns=\"http://www.google.com/transport/p2p\"/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const uint32 kAudioSsrc = 4294967295U;
const uint32 kVideoSsrc = 87654321;
const uint32 kDataSsrc = 1010101;
const uint32 kDataSid = 0;
// Note that this message does not specify a session ID. It must be populated
// before use.
const std::string kGingleAcceptWithSsrcs(
"<iq xmlns='jabber:client' from='me@mydomain.com' " \
" to='user@domain.com/resource' type='set' id='150'> " \
" <session xmlns='http://www.google.com/session' type='accept' " \
" initiator='me@domain.com/resource'> " \
" <description xmlns='http://www.google.com/session/video'> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='103' name='ISAC' clockrate='16000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='104' name='ISAC' clockrate='32000' /> " \
" <src-id xmlns='http://www.google.com/session/phone'> " \
" 4294967295</src-id> " \
" <src-id>87654321</src-id> " \
" </description> " \
" </session> " \
"</iq> ");
const std::string kJingleAcceptWithSsrcs(
"<iq xmlns='jabber:client' from='me@mydomain.com' " \
" to='user@domain.com/resource' type='set' id='150'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-accept' " \
" initiator='me@domain.com/resource'> " \
" <content name='audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='audio' ssrc='4294967295'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" <payload-type id='104' name='ISAC' clockrate='32000'/> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" <content name='video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='video' ssrc='87654321'> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kJingleAcceptWithRtpDataSsrcs(
"<iq xmlns='jabber:client' from='me@mydomain.com' " \
" to='user@domain.com/resource' type='set' id='150'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-accept' " \
" initiator='me@domain.com/resource'> " \
" <content name='audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='audio' ssrc='4294967295'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" <payload-type id='104' name='ISAC' clockrate='32000'/> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" <content name='video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='video' ssrc='87654321'> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" <content name='data'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='data' ssrc='1010101'> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" </jingle> " \
"</iq> ");
const std::string kJingleAcceptWithSctpData(
"<iq xmlns='jabber:client' from='me@mydomain.com' " \
" to='user@domain.com/resource' type='set' id='150'> " \
" <jingle xmlns='urn:xmpp:jingle:1' action='session-accept' " \
" initiator='me@domain.com/resource'> " \
" <content name='audio'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='audio' ssrc='4294967295'> " \
" <payload-type id='103' name='ISAC' clockrate='16000'/> " \
" <payload-type id='104' name='ISAC' clockrate='32000'/> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" <content name='video'> " \
" <description xmlns='urn:xmpp:jingle:apps:rtp:1' " \
" media='video' ssrc='87654321'> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" <content name='data'> " \
" <description xmlns='google:jingle:sctp'> " \
" <stream sid='1'/> " \
" </description> " \
" <transport xmlns='http://www.google.com/transport/p2p'/> " \
" </content> " \
" </jingle> " \
"</iq> ");
std::string JingleView(const std::string& ssrc,
const std::string& width,
const std::string& height,
const std::string& framerate) {
// We have some slightly weird whitespace formatting to make the
// actual XML generated match the expected XML here.
return \
"<cli:iq"
" to='me@mydomain.com'"
" type='set'"
" xmlns:cli='jabber:client'>"
"<jingle"
" xmlns='urn:xmpp:jingle:1'"
" action='session-info'"
" sid=''>"
"<view xmlns='google:jingle'"
" name='video'"
" type='static'"
" ssrc='" + ssrc + "'>"
"<params"
" width='" + width + "'"
" height='" + height + "'"
" framerate='" + framerate + "'"
" preference='0'/>"
"</view>"
"</jingle>"
"</cli:iq>";
}
std::string JingleStreamAdd(const std::string& content_name,
const std::string& nick,
const std::string& name,
const std::string& ssrc) {
return \
"<iq"
" xmlns='jabber:client'"
" from='me@mydomain.com'"
" to='user@domain.com/resource'"
" type='set'"
" id='150'>"
" <jingle"
" xmlns='urn:xmpp:jingle:1'"
" action='description-info'>"
" <content"
" xmlns='urn:xmpp:jingle:1'"
" name='" + content_name + "'>"
" <description"
" xmlns='urn:xmpp:jingle:apps:rtp:1'"
" media='" + content_name + "'>"
" <streams"
" xmlns='google:jingle'>"
" <stream"
" nick='" + nick + "'"
" name='" + name + "'>"
" <ssrc>" + ssrc + "</ssrc>"
" </stream>"
" </streams>"
" </description>"
" </content>"
" </jingle>"
"</iq>";
}
std::string JingleOutboundStreamRemove(const std::string& sid,
const std::string& content_name,
const std::string& name) {
return \
"<cli:iq"
" to='me@mydomain.com'"
" type='set'"
" xmlns:cli='jabber:client'>"
"<jingle"
" xmlns='urn:xmpp:jingle:1'"
" action='description-info'"
" sid='" + sid + "'>"
"<content"
" name='" + content_name + "'"
" creator='initiator'>"
"<description"
" xmlns='urn:xmpp:jingle:apps:rtp:1'"
" media='" + content_name + "'>"
"<streams"
" xmlns='google:jingle'>"
"<stream"
" name='" + name + "'>"
"</stream>"
"</streams>"
"</description>"
"</content>"
"</jingle>"
"</cli:iq>";
}
std::string JingleOutboundStreamAdd(const std::string& sid,
const std::string& content_name,
const std::string& name,
const std::string& ssrc) {
return \
"<cli:iq"
" to='me@mydomain.com'"
" type='set'"
" xmlns:cli='jabber:client'>"
"<jingle"
" xmlns='urn:xmpp:jingle:1'"
" action='description-info'"
" sid='" + sid + "'>"
"<content"
" name='" + content_name + "'"
" creator='initiator'>"
"<description"
" xmlns='urn:xmpp:jingle:apps:rtp:1'"
" media='" + content_name + "'>"
"<streams"
" xmlns='google:jingle'>"
"<stream"
" name='" + name + "'>"
"<ssrc>" + ssrc + "</ssrc>"
"</stream>"
"</streams>"
"</description>"
"</content>"
"</jingle>"
"</cli:iq>";
}
std::string JingleStreamAddWithoutSsrc(const std::string& content_name,
const std::string& nick,
const std::string& name) {
return \
"<iq"
" xmlns='jabber:client'"
" from='me@mydomain.com'"
" to='user@domain.com/resource'"
" type='set'"
" id='150'>"
" <jingle"
" xmlns='urn:xmpp:jingle:1'"
" action='description-info'>"
" <content"
" xmlns='urn:xmpp:jingle:1'"
" name='" + content_name + "'>"
" <description"
" xmlns='urn:xmpp:jingle:apps:rtp:1'"
" media='" + content_name + "'>"
" <streams"
" xmlns='google:jingle'>"
" <stream"
" nick='" + nick + "'"
" name='" + name + "'>"
" </stream>"
" </streams>"
" </description>"
" </content>"
" </jingle>"
"</iq>";
}
std::string JingleStreamRemove(const std::string& content_name,
const std::string& nick,
const std::string& name) {
return \
"<iq"
" xmlns='jabber:client'"
" from='me@mydomain.com'"
" to='user@domain.com/resource'"
" type='set'"
" id='150'>"
" <jingle"
" xmlns='urn:xmpp:jingle:1'"
" action='description-info'>"
" <content"
" xmlns='urn:xmpp:jingle:1'"
" name='" + content_name + "'>"
" <description"
" xmlns='urn:xmpp:jingle:apps:rtp:1'"
" media='" + content_name + "'>"
" <streams"
" xmlns='google:jingle'>"
" <stream"
" nick='" + nick + "'"
" name='" + name + "'/>"
" </streams>"
" </description>"
" </content>"
" </jingle>"
"</iq>";
}
// Convenience function to get CallOptions that have audio enabled,
// but not video or data.
static cricket::CallOptions AudioCallOptions() {
cricket::CallOptions options;
options.has_audio = true;
options.has_video = false;
options.data_channel_type = cricket::DCT_NONE;
return options;
}
// Convenience function to get CallOptions that have audio and video
// enabled, but not data.
static cricket::CallOptions VideoCallOptions() {
cricket::CallOptions options;
options.has_audio = true;
options.has_video = true;
options.data_channel_type = cricket::DCT_NONE;
return options;
}
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static buzz::XmlElement* CopyElement(const buzz::XmlElement* elem) {
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return new buzz::XmlElement(*elem);
}
static std::string AddEncryption(std::string stanza, std::string encryption) {
std::string::size_type pos = stanza.find("</description>");
while (pos != std::string::npos) {
stanza = stanza.insert(pos, encryption);
pos = stanza.find("</description>", pos + encryption.length() + 1);
}
return stanza;
}
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static int IntFromJingleCodecParameter(const buzz::XmlElement* parameter,
const std::string& expected_name) {
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if (parameter) {
const std::string& actual_name =
parameter->Attr(cricket::QN_PAYLOADTYPE_PARAMETER_NAME);
EXPECT_EQ(expected_name, actual_name)
<< "wrong parameter name. Expected '"
<< expected_name << "'. Actually '"
<< actual_name << "'.";
return atoi(parameter->Attr(
cricket::QN_PAYLOADTYPE_PARAMETER_VALUE).c_str());
}
return 0;
}
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template <class CodecClass, class DescriptionClass>
static void VerifyCodecFbParams(const FeedbackParams& expected,
const DescriptionClass* desc) {
if (!expected.params().empty()) {
ASSERT_TRUE(desc != NULL);
const std::vector<CodecClass> codecs = desc->codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
EXPECT_EQ(expected, codecs[i].feedback_params);
}
}
}
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// Parses and extracts payload and codec info from test XML. Since
// that XML will be in various contents (Gingle and Jingle), we need an
// abstract parser with one concrete implementation per XML content.
class MediaSessionTestParser {
public:
virtual buzz::XmlElement* ActionFromStanza(buzz::XmlElement* stanza) = 0;
virtual buzz::XmlElement* ContentFromAction(buzz::XmlElement* action) = 0;
virtual buzz::XmlElement* NextContent(buzz::XmlElement* content) = 0;
virtual buzz::XmlElement* PayloadTypeFromContent(
buzz::XmlElement* content) = 0;
virtual buzz::XmlElement* NextFromPayloadType(
buzz::XmlElement* payload_type) = 0;
virtual cricket::AudioCodec AudioCodecFromPayloadType(
const buzz::XmlElement* payload_type) = 0;
virtual cricket::VideoCodec VideoCodecFromPayloadType(
const buzz::XmlElement* payload_type) = 0;
virtual cricket::DataCodec DataCodecFromPayloadType(
const buzz::XmlElement* payload_type) = 0;
virtual buzz::XmlElement* EncryptionFromContent(
buzz::XmlElement* content) = 0;
virtual buzz::XmlElement* NextFromEncryption(
buzz::XmlElement* encryption) = 0;
virtual const buzz::XmlElement* BandwidthFromContent(
buzz::XmlElement* content) = 0;
virtual const buzz::XmlElement* RtcpMuxFromContent(
buzz::XmlElement* content) = 0;
virtual bool ActionIsTerminate(const buzz::XmlElement* action) = 0;
virtual ~MediaSessionTestParser() {}
};
class JingleSessionTestParser : public MediaSessionTestParser {
public:
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JingleSessionTestParser() {}
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~JingleSessionTestParser() {
}
buzz::XmlElement* ActionFromStanza(buzz::XmlElement* stanza) {
return stanza->FirstNamed(cricket::QN_JINGLE);
}
buzz::XmlElement* ContentFromAction(buzz::XmlElement* action) {
// We need to be able to use multiple contents, but the action
// gets deleted before we can call NextContent, so we need to
// stash away a copy.
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action_.reset(CopyElement(action));
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return action_->FirstNamed(cricket::QN_JINGLE_CONTENT);
}
buzz::XmlElement* NextContent(buzz::XmlElement* content) {
// For some reason, content->NextNamed(cricket::QN_JINGLE_CONTENT)
// doesn't work.
return action_->FirstNamed(cricket::QN_JINGLE_CONTENT)
->NextNamed(cricket::QN_JINGLE_CONTENT);
}
buzz::XmlElement* PayloadTypeFromContent(buzz::XmlElement* content) {
buzz::XmlElement* content_desc =
content->FirstNamed(cricket::QN_JINGLE_RTP_CONTENT);
if (!content_desc)
return NULL;
return content_desc->FirstNamed(cricket::QN_JINGLE_RTP_PAYLOADTYPE);
}
buzz::XmlElement* NextFromPayloadType(buzz::XmlElement* payload_type) {
return payload_type->NextNamed(cricket::QN_JINGLE_RTP_PAYLOADTYPE);
}
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void ParsePayloadTypeFeedbackParameters(const buzz::XmlElement* element,
FeedbackParams* params) {
const buzz::XmlElement* param =
element->FirstNamed(cricket::QN_JINGLE_RTCP_FB);
for (; param != NULL;
param = param->NextNamed(cricket::QN_JINGLE_RTCP_FB)) {
std::string type = param->Attr(cricket::QN_TYPE);
std::string subtype = param->Attr(cricket::QN_SUBTYPE);
if (!type.empty()) {
params->Add(FeedbackParam(type, subtype));
}
}
}
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cricket::AudioCodec AudioCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
int id = 0;
if (payload_type->HasAttr(cricket::QN_ID))
id = atoi(payload_type->Attr(cricket::QN_ID).c_str());
std::string name;
if (payload_type->HasAttr(cricket::QN_NAME))
name = payload_type->Attr(cricket::QN_NAME);
int clockrate = 0;
if (payload_type->HasAttr(cricket::QN_CLOCKRATE))
clockrate = atoi(payload_type->Attr(cricket::QN_CLOCKRATE).c_str());
int bitrate = IntFromJingleCodecParameter(
payload_type->FirstNamed(cricket::QN_PARAMETER), "bitrate");
int channels = 1;
if (payload_type->HasAttr(cricket::QN_CHANNELS))
channels = atoi(payload_type->Attr(
cricket::QN_CHANNELS).c_str());
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AudioCodec codec = AudioCodec(id, name, clockrate, bitrate, channels, 0);
ParsePayloadTypeFeedbackParameters(payload_type, &codec.feedback_params);
return codec;
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}
cricket::VideoCodec VideoCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
int id = 0;
if (payload_type->HasAttr(cricket::QN_ID))
id = atoi(payload_type->Attr(cricket::QN_ID).c_str());
std::string name;
if (payload_type->HasAttr(cricket::QN_NAME))
name = payload_type->Attr(cricket::QN_NAME);
int width = 0;
int height = 0;
int framerate = 0;
const buzz::XmlElement* param =
payload_type->FirstNamed(cricket::QN_PARAMETER);
if (param) {
width = IntFromJingleCodecParameter(param, "width");
param = param->NextNamed(cricket::QN_PARAMETER);
if (param) {
height = IntFromJingleCodecParameter(param, "height");
param = param->NextNamed(cricket::QN_PARAMETER);
if (param) {
framerate = IntFromJingleCodecParameter(param, "framerate");
}
}
}
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VideoCodec codec = VideoCodec(id, name, width, height, framerate, 0);
ParsePayloadTypeFeedbackParameters(payload_type, &codec.feedback_params);
return codec;
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}
cricket::DataCodec DataCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
int id = 0;
if (payload_type->HasAttr(cricket::QN_ID))
id = atoi(payload_type->Attr(cricket::QN_ID).c_str());
std::string name;
if (payload_type->HasAttr(cricket::QN_NAME))
name = payload_type->Attr(cricket::QN_NAME);
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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DataCodec codec = DataCodec(id, name, 0);
ParsePayloadTypeFeedbackParameters(payload_type, &codec.feedback_params);
return codec;
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}
bool ActionIsTerminate(const buzz::XmlElement* action) {
return (action->HasAttr(cricket::QN_ACTION) &&
action->Attr(cricket::QN_ACTION) == "session-terminate");
}
buzz::XmlElement* EncryptionFromContent(buzz::XmlElement* content) {
buzz::XmlElement* content_desc =
content->FirstNamed(cricket::QN_JINGLE_RTP_CONTENT);
if (!content_desc)
return NULL;
return content_desc->FirstNamed(cricket::QN_ENCRYPTION);
}
buzz::XmlElement* NextFromEncryption(buzz::XmlElement* encryption) {
return encryption->NextNamed(cricket::QN_ENCRYPTION);
}
const buzz::XmlElement* BandwidthFromContent(buzz::XmlElement* content) {
buzz::XmlElement* content_desc =
content->FirstNamed(cricket::QN_JINGLE_RTP_CONTENT);
if (!content_desc)
return NULL;
return content_desc->FirstNamed(cricket::QN_JINGLE_RTP_BANDWIDTH);
}
const buzz::XmlElement* RtcpMuxFromContent(buzz::XmlElement* content) {
return content->FirstNamed(cricket::QN_JINGLE_RTCP_MUX);
}
private:
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talk_base::scoped_ptr<buzz::XmlElement> action_;
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};
class GingleSessionTestParser : public MediaSessionTestParser {
public:
GingleSessionTestParser() : found_content_count_(0) {}
buzz::XmlElement* ActionFromStanza(buzz::XmlElement* stanza) {
return stanza->FirstNamed(cricket::QN_GINGLE_SESSION);
}
buzz::XmlElement* ContentFromAction(buzz::XmlElement* session) {
buzz::XmlElement* content =
session->FirstNamed(cricket::QN_GINGLE_AUDIO_CONTENT);
if (content == NULL)
content = session->FirstNamed(cricket::QN_GINGLE_VIDEO_CONTENT);
return content;
}
// Assumes contents are in order of audio, and then video.
buzz::XmlElement* NextContent(buzz::XmlElement* content) {
found_content_count_++;
return content;
}
buzz::XmlElement* PayloadTypeFromContent(buzz::XmlElement* content) {
if (found_content_count_ > 0) {
return content->FirstNamed(cricket::QN_GINGLE_VIDEO_PAYLOADTYPE);
} else {
return content->FirstNamed(cricket::QN_GINGLE_AUDIO_PAYLOADTYPE);
}
}
buzz::XmlElement* NextFromPayloadType(buzz::XmlElement* payload_type) {
if (found_content_count_ > 0) {
return payload_type->NextNamed(cricket::QN_GINGLE_VIDEO_PAYLOADTYPE);
} else {
return payload_type->NextNamed(cricket::QN_GINGLE_AUDIO_PAYLOADTYPE);
}
}
cricket::AudioCodec AudioCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
int id = 0;
if (payload_type->HasAttr(cricket::QN_ID))
id = atoi(payload_type->Attr(cricket::QN_ID).c_str());
std::string name;
if (payload_type->HasAttr(cricket::QN_NAME))
name = payload_type->Attr(cricket::QN_NAME);
int clockrate = 0;
if (payload_type->HasAttr(cricket::QN_CLOCKRATE))
clockrate = atoi(payload_type->Attr(cricket::QN_CLOCKRATE).c_str());
int bitrate = 0;
if (payload_type->HasAttr(cricket::QN_BITRATE))
bitrate = atoi(payload_type->Attr(cricket::QN_BITRATE).c_str());
int channels = 1;
if (payload_type->HasAttr(cricket::QN_CHANNELS))
channels = atoi(payload_type->Attr(cricket::QN_CHANNELS).c_str());
return cricket::AudioCodec(id, name, clockrate, bitrate, channels, 0);
}
cricket::VideoCodec VideoCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
int id = 0;
if (payload_type->HasAttr(cricket::QN_ID))
id = atoi(payload_type->Attr(cricket::QN_ID).c_str());
std::string name;
if (payload_type->HasAttr(cricket::QN_NAME))
name = payload_type->Attr(cricket::QN_NAME);
int width = 0;
if (payload_type->HasAttr(cricket::QN_WIDTH))
width = atoi(payload_type->Attr(cricket::QN_WIDTH).c_str());
int height = 0;
if (payload_type->HasAttr(cricket::QN_HEIGHT))
height = atoi(payload_type->Attr(cricket::QN_HEIGHT).c_str());
int framerate = 1;
if (payload_type->HasAttr(cricket::QN_FRAMERATE))
framerate = atoi(payload_type->Attr(cricket::QN_FRAMERATE).c_str());
return cricket::VideoCodec(id, name, width, height, framerate, 0);
}
cricket::DataCodec DataCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
// Gingle can't do data codecs.
return cricket::DataCodec(0, "", 0);
}
buzz::XmlElement* EncryptionFromContent(
buzz::XmlElement* content) {
return content->FirstNamed(cricket::QN_ENCRYPTION);
}
buzz::XmlElement* NextFromEncryption(buzz::XmlElement* encryption) {
return encryption->NextNamed(cricket::QN_ENCRYPTION);
}
const buzz::XmlElement* BandwidthFromContent(buzz::XmlElement* content) {
return content->FirstNamed(cricket::QN_GINGLE_VIDEO_BANDWIDTH);
}
const buzz::XmlElement* RtcpMuxFromContent(buzz::XmlElement* content) {
return NULL;
}
bool ActionIsTerminate(const buzz::XmlElement* session) {
return (session->HasAttr(buzz::QN_TYPE) &&
session->Attr(buzz::QN_TYPE) == "terminate");
}
int found_content_count_;
};
class MediaSessionClientTest : public sigslot::has_slots<> {
public:
explicit MediaSessionClientTest(MediaSessionTestParser* parser,
cricket::SignalingProtocol initial_protocol) {
nm_ = new talk_base::BasicNetworkManager();
pa_ = new cricket::BasicPortAllocator(nm_);
sm_ = new cricket::SessionManager(pa_, NULL);
fme_ = new cricket::FakeMediaEngine();
fdme_ = new cricket::FakeDataEngine();
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FeedbackParams params_nack_fir;
params_nack_fir.Add(FeedbackParam(cricket::kRtcpFbParamCcm,
cricket::kRtcpFbCcmParamFir));
params_nack_fir.Add(FeedbackParam(cricket::kRtcpFbParamNack));
FeedbackParams params_nack;
params_nack.Add(FeedbackParam(cricket::kRtcpFbParamNack));
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std::vector<cricket::AudioCodec>
audio_codecs(kAudioCodecs, kAudioCodecs + ARRAY_SIZE(kAudioCodecs));
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SetCodecFeedbackParams(&audio_codecs, params_nack);
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fme_->SetAudioCodecs(audio_codecs);
std::vector<cricket::VideoCodec>
video_codecs(kVideoCodecs, kVideoCodecs + ARRAY_SIZE(kVideoCodecs));
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SetCodecFeedbackParams(&video_codecs, params_nack_fir);
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fme_->SetVideoCodecs(video_codecs);
std::vector<cricket::DataCodec>
data_codecs(kDataCodecs, kDataCodecs + ARRAY_SIZE(kDataCodecs));
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SetCodecFeedbackParams(&data_codecs, params_nack);
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fdme_->SetDataCodecs(data_codecs);
client_ = new cricket::MediaSessionClient(
buzz::Jid("user@domain.com/resource"), sm_,
fme_, fdme_, new cricket::FakeDeviceManager());
client_->session_manager()->SignalOutgoingMessage.connect(
this, &MediaSessionClientTest::OnSendStanza);
client_->session_manager()->SignalSessionCreate.connect(
this, &MediaSessionClientTest::OnSessionCreate);
client_->SignalCallCreate.connect(
this, &MediaSessionClientTest::OnCallCreate);
client_->SignalCallDestroy.connect(
this, &MediaSessionClientTest::OnCallDestroy);
call_ = NULL;
parser_ = parser;
initial_protocol_ = initial_protocol;
expect_incoming_crypto_ = false;
expect_outgoing_crypto_ = false;
expected_video_bandwidth_ = cricket::kAutoBandwidth;
expected_video_rtcp_mux_ = false;
}
~MediaSessionClientTest() {
delete client_;
delete sm_;
delete pa_;
delete nm_;
delete parser_;
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ClearStanzas();
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}
buzz::XmlElement* ActionFromStanza(buzz::XmlElement* stanza) {
return parser_->ActionFromStanza(stanza);
}
buzz::XmlElement* ContentFromAction(buzz::XmlElement* action) {
return parser_->ContentFromAction(action);
}
buzz::XmlElement* PayloadTypeFromContent(buzz::XmlElement* payload) {
return parser_->PayloadTypeFromContent(payload);
}
buzz::XmlElement* NextFromPayloadType(buzz::XmlElement* payload_type) {
return parser_->NextFromPayloadType(payload_type);
}
buzz::XmlElement* EncryptionFromContent(buzz::XmlElement* content) {
return parser_->EncryptionFromContent(content);
}
buzz::XmlElement* NextFromEncryption(buzz::XmlElement* encryption) {
return parser_->NextFromEncryption(encryption);
}
cricket::AudioCodec AudioCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
return parser_->AudioCodecFromPayloadType(payload_type);
}
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cricket::VideoCodec VideoCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
return parser_->VideoCodecFromPayloadType(payload_type);
}
cricket::DataCodec DataCodecFromPayloadType(
const buzz::XmlElement* payload_type) {
return parser_->DataCodecFromPayloadType(payload_type);
}
const AudioContentDescription* GetFirstAudioContentDescription(
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const cricket::SessionDescription* sdesc) {
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(sdesc);
if (content == NULL)
return NULL;
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return static_cast<const AudioContentDescription*>(content->description);
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}
const cricket::VideoContentDescription* GetFirstVideoContentDescription(
const cricket::SessionDescription* sdesc) {
const cricket::ContentInfo* content =
cricket::GetFirstVideoContent(sdesc);
if (content == NULL)
return NULL;
return static_cast<const cricket::VideoContentDescription*>(
content->description);
}
void CheckCryptoFromGoodIncomingInitiate(const cricket::Session* session) {
ASSERT_TRUE(session != NULL);
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const AudioContentDescription* content =
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GetFirstAudioContentDescription(session->remote_description());
ASSERT_TRUE(content != NULL);
ASSERT_EQ(2U, content->cryptos().size());
ASSERT_EQ(145, content->cryptos()[0].tag);
ASSERT_EQ("AES_CM_128_HMAC_SHA1_32", content->cryptos()[0].cipher_suite);
ASSERT_EQ("inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9",
content->cryptos()[0].key_params);
ASSERT_EQ(51, content->cryptos()[1].tag);
ASSERT_EQ("AES_CM_128_HMAC_SHA1_80", content->cryptos()[1].cipher_suite);
ASSERT_EQ("inline:J4lfdUL8W1F7TNJKcbuygaQuA429SJy2e9JctPUy",
content->cryptos()[1].key_params);
}
void CheckCryptoForGoodOutgoingAccept(const cricket::Session* session) {
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const AudioContentDescription* content =
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GetFirstAudioContentDescription(session->local_description());
ASSERT_EQ(1U, content->cryptos().size());
ASSERT_EQ(145, content->cryptos()[0].tag);
ASSERT_EQ("AES_CM_128_HMAC_SHA1_32", content->cryptos()[0].cipher_suite);
ASSERT_EQ(47U, content->cryptos()[0].key_params.size());
}
void CheckBadCryptoFromIncomingInitiate(const cricket::Session* session) {
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const AudioContentDescription* content =
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GetFirstAudioContentDescription(session->remote_description());
ASSERT_EQ(1U, content->cryptos().size());
ASSERT_EQ(145, content->cryptos()[0].tag);
ASSERT_EQ("NOT_SUPPORTED", content->cryptos()[0].cipher_suite);
ASSERT_EQ("inline:hsWuSQJxx7przmb8HM+ZkeNcG3HezSNID7LmfDa9",
content->cryptos()[0].key_params);
}
void CheckNoCryptoForOutgoingAccept(const cricket::Session* session) {
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const AudioContentDescription* content =
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GetFirstAudioContentDescription(session->local_description());
ASSERT_TRUE(content->cryptos().empty());
}
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void CheckRtcpFb(const cricket::SessionDescription* sdesc) {
VerifyCodecFbParams<AudioCodec>(expected_audio_fb_params_,
GetFirstAudioContentDescription(sdesc));
VerifyCodecFbParams<VideoCodec>(expected_video_fb_params_,
GetFirstVideoContentDescription(sdesc));
VerifyCodecFbParams<DataCodec>(expected_data_fb_params_,
GetFirstDataContentDescription(sdesc));
}
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void CheckVideoBandwidth(int expected_bandwidth,
const cricket::SessionDescription* sdesc) {
const cricket::VideoContentDescription* video =
GetFirstVideoContentDescription(sdesc);
if (video != NULL) {
ASSERT_EQ(expected_bandwidth, video->bandwidth());
}
}
void CheckVideoRtcpMux(bool expected_video_rtcp_mux,
const cricket::SessionDescription* sdesc) {
const cricket::VideoContentDescription* video =
GetFirstVideoContentDescription(sdesc);
if (video != NULL) {
ASSERT_EQ(expected_video_rtcp_mux, video->rtcp_mux());
}
}
virtual void CheckRtpDataContent(buzz::XmlElement* content) {
if (initial_protocol_) {
// Gingle can not write out data content.
return;
}
buzz::XmlElement* e = PayloadTypeFromContent(content);
ASSERT_TRUE(e != NULL);
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cricket::DataCodec codec = DataCodecFromPayloadType(e);
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EXPECT_EQ(127, codec.id);
EXPECT_EQ("google-data", codec.name);
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EXPECT_EQ(expected_data_fb_params_, codec.feedback_params);
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CheckDataRtcpMux(true, call_->sessions()[0]->local_description());
CheckDataRtcpMux(true, call_->sessions()[0]->remote_description());
if (expect_outgoing_crypto_) {
content = parser_->NextContent(content);
buzz::XmlElement* encryption = EncryptionFromContent(content);
ASSERT_TRUE(encryption != NULL);
// TODO(pthatcher): Check encryption parameters?
}
}
virtual void CheckSctpDataContent(buzz::XmlElement* content) {
if (initial_protocol_) {
// Gingle can not write out data content.
return;
}
buzz::XmlElement* payload_type = PayloadTypeFromContent(content);
ASSERT_TRUE(payload_type == NULL);
buzz::XmlElement* encryption = EncryptionFromContent(content);
ASSERT_TRUE(encryption == NULL);
// TODO(pthatcher): Check for <streams>.
}
void CheckDataRtcpMux(bool expected_data_rtcp_mux,
const cricket::SessionDescription* sdesc) {
const cricket::DataContentDescription* data =
GetFirstDataContentDescription(sdesc);
if (data != NULL) {
ASSERT_EQ(expected_data_rtcp_mux, data->rtcp_mux());
}
}
void CheckAudioSsrcForIncomingAccept(const cricket::Session* session) {
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const AudioContentDescription* audio =
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GetFirstAudioContentDescription(session->remote_description());
ASSERT_TRUE(audio != NULL);
ASSERT_EQ(kAudioSsrc, audio->first_ssrc());
}
void CheckVideoSsrcForIncomingAccept(const cricket::Session* session) {
const cricket::VideoContentDescription* video =
GetFirstVideoContentDescription(session->remote_description());
ASSERT_TRUE(video != NULL);
ASSERT_EQ(kVideoSsrc, video->first_ssrc());
}
void CheckDataSsrcForIncomingAccept(const cricket::Session* session) {
const cricket::DataContentDescription* data =
GetFirstDataContentDescription(session->remote_description());
ASSERT_TRUE(data != NULL);
ASSERT_EQ(kDataSsrc, data->first_ssrc());
}
void TestGoodIncomingInitiate(const std::string& initiate_string,
const cricket::CallOptions& options,
buzz::XmlElement** element) {
*element = NULL;
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talk_base::scoped_ptr<buzz::XmlElement> el(
buzz::XmlElement::ForStr(initiate_string));
client_->session_manager()->OnIncomingMessage(el.get());
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ASSERT_TRUE(call_ != NULL);
ASSERT_TRUE(call_->sessions()[0] != NULL);
ASSERT_EQ(cricket::Session::STATE_RECEIVEDINITIATE,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_RESULT), stanzas_[0]->Attr(buzz::QN_TYPE));
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ClearStanzas();
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CheckVideoBandwidth(expected_video_bandwidth_,
call_->sessions()[0]->remote_description());
CheckVideoRtcpMux(expected_video_rtcp_mux_,
call_->sessions()[0]->remote_description());
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CheckRtcpFb(call_->sessions()[0]->remote_description());
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if (expect_incoming_crypto_) {
CheckCryptoFromGoodIncomingInitiate(call_->sessions()[0]);
}
// TODO(pthatcher): Add tests for sending <bandwidth> in accept.
call_->AcceptSession(call_->sessions()[0], options);
ASSERT_EQ(cricket::Session::STATE_SENTACCEPT,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_SET), stanzas_[0]->Attr(buzz::QN_TYPE));
buzz::XmlElement* e = ActionFromStanza(stanzas_[0]);
ASSERT_TRUE(e != NULL);
ASSERT_TRUE(ContentFromAction(e) != NULL);
*element = CopyElement(ContentFromAction(e));
ASSERT_TRUE(*element != NULL);
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ClearStanzas();
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if (expect_outgoing_crypto_) {
CheckCryptoForGoodOutgoingAccept(call_->sessions()[0]);
}
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if (options.data_channel_type == cricket::DCT_RTP) {
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CheckDataRtcpMux(true, call_->sessions()[0]->local_description());
CheckDataRtcpMux(true, call_->sessions()[0]->remote_description());
// TODO(pthatcher): Check rtcpmux and crypto?
}
call_->Terminate();
ASSERT_EQ(cricket::Session::STATE_SENTTERMINATE,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_SET), stanzas_[0]->Attr(buzz::QN_TYPE));
e = ActionFromStanza(stanzas_[0]);
ASSERT_TRUE(e != NULL);
ASSERT_TRUE(parser_->ActionIsTerminate(e));
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ClearStanzas();
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}
void TestRejectOffer(const std::string &initiate_string,
const cricket::CallOptions& options,
buzz::XmlElement** element) {
*element = NULL;
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talk_base::scoped_ptr<buzz::XmlElement> el(
buzz::XmlElement::ForStr(initiate_string));
client_->session_manager()->OnIncomingMessage(el.get());
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ASSERT_TRUE(call_ != NULL);
ASSERT_TRUE(call_->sessions()[0] != NULL);
ASSERT_EQ(cricket::Session::STATE_RECEIVEDINITIATE,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_RESULT), stanzas_[0]->Attr(buzz::QN_TYPE));
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ClearStanzas();
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call_->AcceptSession(call_->sessions()[0], options);
ASSERT_EQ(cricket::Session::STATE_SENTACCEPT,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_SET), stanzas_[0]->Attr(buzz::QN_TYPE));
buzz::XmlElement* e = ActionFromStanza(stanzas_[0]);
ASSERT_TRUE(e != NULL);
ASSERT_TRUE(ContentFromAction(e) != NULL);
*element = CopyElement(ContentFromAction(e));
ASSERT_TRUE(*element != NULL);
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ClearStanzas();
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buzz::XmlElement* content = *element;
// The NextContent method actually returns the second content. So we
// can't handle the case when audio, video and data are all enabled. But
// since we are testing rejection, it won't be the case.
if (options.has_audio) {
ASSERT_TRUE(content != NULL);
ASSERT_EQ("test audio", content->Attr(buzz::QName("", "name")));
content = parser_->NextContent(content);
}
if (options.has_video) {
ASSERT_TRUE(content != NULL);
ASSERT_EQ("test video", content->Attr(buzz::QName("", "name")));
content = parser_->NextContent(content);
}
if (options.has_data()) {
ASSERT_TRUE(content != NULL);
ASSERT_EQ("test data", content->Attr(buzz::QName("", "name")));
content = parser_->NextContent(content);
}
call_->Terminate();
ASSERT_EQ(cricket::Session::STATE_SENTTERMINATE,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_SET), stanzas_[0]->Attr(buzz::QN_TYPE));
e = ActionFromStanza(stanzas_[0]);
ASSERT_TRUE(e != NULL);
ASSERT_TRUE(parser_->ActionIsTerminate(e));
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ClearStanzas();
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}
void TestBadIncomingInitiate(const std::string& initiate_string) {
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talk_base::scoped_ptr<buzz::XmlElement> el(
buzz::XmlElement::ForStr(initiate_string));
client_->session_manager()->OnIncomingMessage(el.get());
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ASSERT_TRUE(call_ != NULL);
ASSERT_TRUE(call_->sessions()[0] != NULL);
ASSERT_EQ(cricket::Session::STATE_SENTREJECT,
call_->sessions()[0]->state());
ASSERT_EQ(2U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[1]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_RESULT), stanzas_[1]->Attr(buzz::QN_TYPE));
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ClearStanzas();
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}
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void VerifyAudioCodec(const AudioCodec& codec, int id,
const std::string& name, int clockrate,
int bitrate, int channels) {
ASSERT_EQ(id, codec.id);
ASSERT_EQ(name, codec.name);
ASSERT_EQ(clockrate, codec.clockrate);
ASSERT_EQ(bitrate, codec.bitrate);
ASSERT_EQ(channels, codec.channels);
ASSERT_EQ(expected_audio_fb_params_, codec.feedback_params);
}
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void TestGoodOutgoingInitiate(const cricket::CallOptions& options) {
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if (initial_protocol_ == cricket::PROTOCOL_JINGLE) {
// rtcp fb is only implemented for jingle.
ExpectRtcpFb();
}
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client_->CreateCall();
ASSERT_TRUE(call_ != NULL);
call_->InitiateSession(buzz::Jid("me@mydomain.com"),
buzz::Jid("me@mydomain.com"), options);
ASSERT_TRUE(call_->sessions()[0] != NULL);
ASSERT_EQ(cricket::Session::STATE_SENTINITIATE,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_SET), stanzas_[0]->Attr(buzz::QN_TYPE));
buzz::XmlElement* action = ActionFromStanza(stanzas_[0]);
ASSERT_TRUE(action != NULL);
buzz::XmlElement* content = ContentFromAction(action);
ASSERT_TRUE(content != NULL);
buzz::XmlElement* e = PayloadTypeFromContent(content);
ASSERT_TRUE(e != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 103, "ISAC", 16000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 104, "ISAC", 32000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 119, "ISACLC", 16000, 40000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 99, "speex", 16000, 22000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 97, "IPCMWB", 16000, 80000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 9, "G722", 16000, 64000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 102, "iLBC", 8000, 13300, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 98, "speex", 8000, 11000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 3, "GSM", 8000, 13000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 100, "EG711U", 8000, 64000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 101, "EG711A", 8000, 64000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 0, "PCMU", 8000, 64000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
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VerifyAudioCodec(codec, 8, "PCMA", 8000, 64000, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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VerifyAudioCodec(codec, 126, "CN", 32000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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VerifyAudioCodec(codec, 105, "CN", 16000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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VerifyAudioCodec(codec, 13, "CN", 8000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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VerifyAudioCodec(codec, 117, "red", 8000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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VerifyAudioCodec(codec, 106, "telephone-event", 8000, 0, 1);
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e = NextFromPayloadType(e);
ASSERT_TRUE(e == NULL);
if (expect_outgoing_crypto_) {
buzz::XmlElement* encryption = EncryptionFromContent(content);
ASSERT_TRUE(encryption != NULL);
if (client_->secure() == cricket::SEC_REQUIRED) {
ASSERT_TRUE(cricket::GetXmlAttr(
encryption, cricket::QN_ENCRYPTION_REQUIRED, false));
}
if (content->Name().Namespace() == cricket::NS_GINGLE_AUDIO) {
e = encryption->FirstNamed(cricket::QN_GINGLE_AUDIO_CRYPTO_USAGE);
ASSERT_TRUE(e != NULL);
ASSERT_TRUE(
e->NextNamed(cricket::QN_GINGLE_AUDIO_CRYPTO_USAGE) == NULL);
ASSERT_TRUE(
e->FirstNamed(cricket::QN_GINGLE_VIDEO_CRYPTO_USAGE) == NULL);
}
e = encryption->FirstNamed(cricket::QN_CRYPTO);
ASSERT_TRUE(e != NULL);
ASSERT_EQ("0", e->Attr(cricket::QN_CRYPTO_TAG));
ASSERT_EQ("AES_CM_128_HMAC_SHA1_32", e->Attr(cricket::QN_CRYPTO_SUITE));
std::string key_0 = e->Attr(cricket::QN_CRYPTO_KEY_PARAMS);
ASSERT_EQ(47U, key_0.length());
ASSERT_EQ("inline:", key_0.substr(0, 7));
e = e->NextNamed(cricket::QN_CRYPTO);
ASSERT_TRUE(e != NULL);
ASSERT_EQ("1", e->Attr(cricket::QN_CRYPTO_TAG));
ASSERT_EQ("AES_CM_128_HMAC_SHA1_80", e->Attr(cricket::QN_CRYPTO_SUITE));
std::string key_1 = e->Attr(cricket::QN_CRYPTO_KEY_PARAMS);
ASSERT_EQ(47U, key_1.length());
ASSERT_EQ("inline:", key_1.substr(0, 7));
ASSERT_NE(key_0, key_1);
encryption = NextFromEncryption(encryption);
ASSERT_TRUE(encryption == NULL);
}
if (options.has_video) {
CheckVideoBandwidth(options.video_bandwidth,
call_->sessions()[0]->local_description());
CheckVideoRtcpMux(expected_video_rtcp_mux_,
call_->sessions()[0]->remote_description());
content = parser_->NextContent(content);
const buzz::XmlElement* bandwidth =
parser_->BandwidthFromContent(content);
if (options.video_bandwidth == cricket::kAutoBandwidth) {
ASSERT_TRUE(bandwidth == NULL);
} else {
ASSERT_TRUE(bandwidth != NULL);
ASSERT_EQ("AS", bandwidth->Attr(buzz::QName("", "type")));
ASSERT_EQ(talk_base::ToString(options.video_bandwidth / 1000),
bandwidth->BodyText());
}
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buzz::XmlElement* e = PayloadTypeFromContent(content);
ASSERT_TRUE(e != NULL);
VideoCodec codec = VideoCodecFromPayloadType(e);
VideoCodec expected_codec = kVideoCodecs[0];
expected_codec.preference = codec.preference;
expected_codec.feedback_params = expected_video_fb_params_;
EXPECT_EQ(expected_codec, codec);
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}
if (options.data_channel_type == cricket::DCT_RTP) {
content = parser_->NextContent(content);
CheckRtpDataContent(content);
}
if (options.data_channel_type == cricket::DCT_SCTP) {
content = parser_->NextContent(content);
CheckSctpDataContent(content);
}
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ClearStanzas();
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}
void TestHasAllSupportedAudioCodecs(buzz::XmlElement* e) {
ASSERT_TRUE(e != NULL);
e = PayloadTypeFromContent(e);
ASSERT_TRUE(e != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(103, codec.id);
ASSERT_EQ("ISAC", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(104, codec.id);
ASSERT_EQ("ISAC", codec.name);
ASSERT_EQ(32000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(119, codec.id);
ASSERT_EQ("ISACLC", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(40000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(99, codec.id);
ASSERT_EQ("speex", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(22000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(97, codec.id);
ASSERT_EQ("IPCMWB", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(80000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(9, codec.id);
ASSERT_EQ("G722", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(64000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(102, codec.id);
ASSERT_EQ("iLBC", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(13300, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(98, codec.id);
ASSERT_EQ("speex", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(11000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(3, codec.id);
ASSERT_EQ("GSM", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(13000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(100, codec.id);
ASSERT_EQ("EG711U", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(64000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(101, codec.id);
ASSERT_EQ("EG711A", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(64000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(0, codec.id);
ASSERT_EQ("PCMU", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(64000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(8, codec.id);
ASSERT_EQ("PCMA", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(64000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(126, codec.id);
ASSERT_EQ("CN", codec.name);
ASSERT_EQ(32000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(105, codec.id);
ASSERT_EQ("CN", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(13, codec.id);
ASSERT_EQ("CN", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(117, codec.id);
ASSERT_EQ("red", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(106, codec.id);
ASSERT_EQ("telephone-event", codec.name);
ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e == NULL);
}
void TestCodecsOfVideoInitiate(buzz::XmlElement* content) {
ASSERT_TRUE(content != NULL);
buzz::XmlElement* payload_type = PayloadTypeFromContent(content);
ASSERT_TRUE(payload_type != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(payload_type);
ASSERT_EQ(103, codec.id);
ASSERT_EQ("ISAC", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
content = parser_->NextContent(content);
ASSERT_TRUE(content != NULL);
payload_type = PayloadTypeFromContent(content);
ASSERT_TRUE(payload_type != NULL);
cricket::VideoCodec vcodec =
parser_->VideoCodecFromPayloadType(payload_type);
ASSERT_EQ(99, vcodec.id);
ASSERT_EQ("H264-SVC", vcodec.name);
ASSERT_EQ(320, vcodec.width);
ASSERT_EQ(200, vcodec.height);
ASSERT_EQ(30, vcodec.framerate);
}
void TestHasAudioCodecsFromInitiateSomeUnsupported(buzz::XmlElement* e) {
ASSERT_TRUE(e != NULL);
e = PayloadTypeFromContent(e);
ASSERT_TRUE(e != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(103, codec.id);
ASSERT_EQ("ISAC", codec.name);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(100, codec.id);
ASSERT_EQ("EG711U", codec.name);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(101, codec.id);
ASSERT_EQ("EG711A", codec.name);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(0, codec.id);
ASSERT_EQ("PCMU", codec.name);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(13, codec.id);
ASSERT_EQ("CN", codec.name);
e = NextFromPayloadType(e);
ASSERT_TRUE(e == NULL);
}
void TestHasAudioCodecsFromInitiateDynamicAudioCodecs(
buzz::XmlElement* e) {
ASSERT_TRUE(e != NULL);
e = PayloadTypeFromContent(e);
ASSERT_TRUE(e != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(123, codec.id);
ASSERT_EQ(16000, codec.clockrate);
ASSERT_EQ(1, codec.channels);
e = NextFromPayloadType(e);
ASSERT_TRUE(e == NULL);
}
void TestHasDefaultAudioCodecs(buzz::XmlElement* e) {
ASSERT_TRUE(e != NULL);
e = PayloadTypeFromContent(e);
ASSERT_TRUE(e != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(103, codec.id);
ASSERT_EQ("ISAC", codec.name);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(0, codec.id);
ASSERT_EQ("PCMU", codec.name);
e = NextFromPayloadType(e);
ASSERT_TRUE(e == NULL);
}
void TestHasAudioCodecsFromInitiateStaticAudioCodecs(
buzz::XmlElement* e) {
ASSERT_TRUE(e != NULL);
e = PayloadTypeFromContent(e);
ASSERT_TRUE(e != NULL);
cricket::AudioCodec codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(3, codec.id);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(0, codec.id);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(8, codec.id);
e = NextFromPayloadType(e);
ASSERT_TRUE(e == NULL);
}
void TestGingleInitiateWithUnsupportedCrypto(
const std::string &initiate_string,
buzz::XmlElement** element) {
*element = NULL;
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talk_base::scoped_ptr<buzz::XmlElement> el(
buzz::XmlElement::ForStr(initiate_string));
client_->session_manager()->OnIncomingMessage(el.get());
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ASSERT_EQ(cricket::Session::STATE_RECEIVEDINITIATE,
call_->sessions()[0]->state());
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ClearStanzas();
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CheckBadCryptoFromIncomingInitiate(call_->sessions()[0]);
call_->AcceptSession(call_->sessions()[0], cricket::CallOptions());
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ClearStanzas();
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CheckNoCryptoForOutgoingAccept(call_->sessions()[0]);
call_->Terminate();
ASSERT_EQ(cricket::Session::STATE_SENTTERMINATE,
call_->sessions()[0]->state());
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ClearStanzas();
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}
void TestIncomingAcceptWithSsrcs(
const std::string& accept_string,
cricket::CallOptions& options) {
client_->CreateCall();
ASSERT_TRUE(call_ != NULL);
call_->InitiateSession(buzz::Jid("me@mydomain.com"),
buzz::Jid("me@mydomain.com"), options);
ASSERT_TRUE(call_->sessions()[0] != NULL);
ASSERT_EQ(cricket::Session::STATE_SENTINITIATE,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_SET), stanzas_[0]->Attr(buzz::QN_TYPE));
buzz::XmlElement* action = ActionFromStanza(stanzas_[0]);
ASSERT_TRUE(action != NULL);
buzz::XmlElement* content = ContentFromAction(action);
ASSERT_TRUE(content != NULL);
if (initial_protocol_ == cricket::PROTOCOL_JINGLE) {
buzz::XmlElement* content_desc =
content->FirstNamed(cricket::QN_JINGLE_RTP_CONTENT);
ASSERT_TRUE(content_desc != NULL);
ASSERT_EQ("", content_desc->Attr(cricket::QN_SSRC));
}
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ClearStanzas();
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// We need to insert the session ID into the session accept message.
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talk_base::scoped_ptr<buzz::XmlElement> el(
buzz::XmlElement::ForStr(accept_string));
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const std::string sid = call_->sessions()[0]->id();
if (initial_protocol_ == cricket::PROTOCOL_JINGLE) {
buzz::XmlElement* jingle = el->FirstNamed(cricket::QN_JINGLE);
jingle->SetAttr(cricket::QN_SID, sid);
} else {
buzz::XmlElement* session = el->FirstNamed(cricket::QN_GINGLE_SESSION);
session->SetAttr(cricket::QN_ID, sid);
}
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client_->session_manager()->OnIncomingMessage(el.get());
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ASSERT_EQ(cricket::Session::STATE_RECEIVEDACCEPT,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_TRUE(buzz::QN_IQ == stanzas_[0]->Name());
ASSERT_TRUE(stanzas_[0]->HasAttr(buzz::QN_TYPE));
ASSERT_EQ(std::string(buzz::STR_RESULT), stanzas_[0]->Attr(buzz::QN_TYPE));
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ClearStanzas();
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CheckAudioSsrcForIncomingAccept(call_->sessions()[0]);
CheckVideoSsrcForIncomingAccept(call_->sessions()[0]);
if (options.data_channel_type == cricket::DCT_RTP) {
CheckDataSsrcForIncomingAccept(call_->sessions()[0]);
}
// TODO(pthatcher): Check kDataSid if DCT_SCTP.
}
size_t ClearStanzas() {
size_t size = stanzas_.size();
for (size_t i = 0; i < size; i++) {
delete stanzas_[i];
}
stanzas_.clear();
return size;
}
buzz::XmlElement* SetJingleSid(buzz::XmlElement* stanza) {
buzz::XmlElement* jingle =
stanza->FirstNamed(cricket::QN_JINGLE);
jingle->SetAttr(cricket::QN_SID, call_->sessions()[0]->id());
return stanza;
}
void TestSendVideoStreamUpdate() {
cricket::CallOptions options = VideoCallOptions();
options.is_muc = true;
client_->CreateCall();
call_->InitiateSession(buzz::Jid("me@mydomain.com"),
buzz::Jid("me@mydomain.com"), options);
ClearStanzas();
cricket::StreamParams stream;
stream.id = "test-stream";
stream.ssrcs.push_back(1001);
talk_base::scoped_ptr<buzz::XmlElement> expected_stream_add(
buzz::XmlElement::ForStr(
JingleOutboundStreamAdd(
call_->sessions()[0]->id(),
"video", stream.id, "1001")));
talk_base::scoped_ptr<buzz::XmlElement> expected_stream_remove(
buzz::XmlElement::ForStr(
JingleOutboundStreamRemove(
call_->sessions()[0]->id(),
"video", stream.id)));
call_->SendVideoStreamUpdate(call_->sessions()[0],
call_->CreateVideoStreamUpdate(stream));
ASSERT_EQ(1U, stanzas_.size());
EXPECT_EQ(expected_stream_add->Str(), stanzas_[0]->Str());
ClearStanzas();
stream.ssrcs.clear();
call_->SendVideoStreamUpdate(call_->sessions()[0],
call_->CreateVideoStreamUpdate(stream));
ASSERT_EQ(1U, stanzas_.size());
EXPECT_EQ(expected_stream_remove->Str(), stanzas_[0]->Str());
ClearStanzas();
}
void TestStreamsUpdateAndViewRequests() {
cricket::CallOptions options = VideoCallOptions();
options.is_muc = true;
client_->CreateCall();
call_->InitiateSession(buzz::Jid("me@mydomain.com"),
buzz::Jid("me@mydomain.com"), options);
ASSERT_EQ(1U, ClearStanzas());
ASSERT_EQ(0U, last_streams_added_.audio().size());
ASSERT_EQ(0U, last_streams_added_.video().size());
ASSERT_EQ(0U, last_streams_removed_.audio().size());
ASSERT_EQ(0U, last_streams_removed_.video().size());
talk_base::scoped_ptr<buzz::XmlElement> accept_stanza(
buzz::XmlElement::ForStr(kJingleAcceptWithSsrcs));
SetJingleSid(accept_stanza.get());
client_->session_manager()->OnIncomingMessage(accept_stanza.get());
ASSERT_EQ(cricket::Session::STATE_RECEIVEDACCEPT,
call_->sessions()[0]->state());
ASSERT_EQ(1U, stanzas_.size());
ASSERT_EQ(std::string(buzz::STR_RESULT), stanzas_[0]->Attr(buzz::QN_TYPE));
ClearStanzas();
// Need to clear the added streams, because they are populated when
// receiving an accept message now.
last_streams_added_.mutable_video()->clear();
last_streams_added_.mutable_audio()->clear();
call_->sessions()[0]->SetState(cricket::Session::STATE_INPROGRESS);
talk_base::scoped_ptr<buzz::XmlElement> streams_stanza(
buzz::XmlElement::ForStr(
JingleStreamAdd("video", "Bob", "video1", "ABC")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
// First one is ignored because of bad syntax.
ASSERT_EQ(1U, stanzas_.size());
// TODO(pthatcher): Figure out how to make this an ERROR rather than RESULT.
ASSERT_EQ(std::string(buzz::STR_ERROR), stanzas_[0]->Attr(buzz::QN_TYPE));
ClearStanzas();
ASSERT_EQ(0U, last_streams_added_.audio().size());
ASSERT_EQ(0U, last_streams_added_.video().size());
ASSERT_EQ(0U, last_streams_removed_.audio().size());
ASSERT_EQ(0U, last_streams_removed_.video().size());
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAdd("audio", "Bob", "audio1", "1234")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_added_.audio().size());
ASSERT_EQ("Bob", last_streams_added_.audio()[0].groupid);
ASSERT_EQ(1U, last_streams_added_.audio()[0].ssrcs.size());
ASSERT_EQ(1234U, last_streams_added_.audio()[0].first_ssrc());
// Ignores adds without ssrcs.
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAddWithoutSsrc("audio", "Bob", "audioX")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_added_.audio().size());
ASSERT_EQ(1234U, last_streams_added_.audio()[0].first_ssrc());
// Ignores stream updates with unknown content names. (Don't terminate).
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAddWithoutSsrc("foo", "Bob", "foo")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAdd("audio", "Joe", "audio1", "2468")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_added_.audio().size());
ASSERT_EQ("Joe", last_streams_added_.audio()[0].groupid);
ASSERT_EQ(1U, last_streams_added_.audio()[0].ssrcs.size());
ASSERT_EQ(2468U, last_streams_added_.audio()[0].first_ssrc());
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAdd("video", "Bob", "video1", "5678")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_added_.video().size());
ASSERT_EQ("Bob", last_streams_added_.video()[0].groupid);
ASSERT_EQ(1U, last_streams_added_.video()[0].ssrcs.size());
ASSERT_EQ(5678U, last_streams_added_.video()[0].first_ssrc());
// We're testing that a "duplicate" is effectively ignored.
last_streams_added_.mutable_video()->clear();
last_streams_removed_.mutable_video()->clear();
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAdd("video", "Bob", "video1", "5678")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(0U, last_streams_added_.video().size());
ASSERT_EQ(0U, last_streams_removed_.video().size());
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamAdd("video", "Bob", "video2", "5679")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_added_.video().size());
ASSERT_EQ("Bob", last_streams_added_.video()[0].groupid);
ASSERT_EQ(1U, last_streams_added_.video()[0].ssrcs.size());
ASSERT_EQ(5679U, last_streams_added_.video()[0].first_ssrc());
cricket::FakeVoiceMediaChannel* voice_channel = fme_->GetVoiceChannel(0);
ASSERT_TRUE(voice_channel != NULL);
ASSERT_TRUE(voice_channel->HasRecvStream(1234U));
ASSERT_TRUE(voice_channel->HasRecvStream(2468U));
cricket::FakeVideoMediaChannel* video_channel = fme_->GetVideoChannel(0);
ASSERT_TRUE(video_channel != NULL);
ASSERT_TRUE(video_channel->HasRecvStream(5678U));
ClearStanzas();
cricket::ViewRequest viewRequest;
cricket::StaticVideoView staticVideoView(
cricket::StreamSelector(5678U), 640, 480, 30);
viewRequest.static_video_views.push_back(staticVideoView);
talk_base::scoped_ptr<buzz::XmlElement> expected_view_elem(
buzz::XmlElement::ForStr(JingleView("5678", "640", "480", "30")));
SetJingleSid(expected_view_elem.get());
ASSERT_TRUE(
call_->SendViewRequest(call_->sessions()[0], viewRequest));
ASSERT_EQ(1U, stanzas_.size());
ASSERT_EQ(expected_view_elem->Str(), stanzas_[0]->Str());
ClearStanzas();
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamRemove("audio", "Bob", "audio1")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_removed_.audio().size());
ASSERT_EQ(1U, last_streams_removed_.audio()[0].ssrcs.size());
EXPECT_EQ(1234U, last_streams_removed_.audio()[0].first_ssrc());
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamRemove("video", "Bob", "video1")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_removed_.video().size());
ASSERT_EQ(1U, last_streams_removed_.video()[0].ssrcs.size());
EXPECT_EQ(5678U, last_streams_removed_.video()[0].first_ssrc());
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamRemove("video", "Bob", "video2")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(1U, last_streams_removed_.video().size());
ASSERT_EQ(1U, last_streams_removed_.video()[0].ssrcs.size());
EXPECT_EQ(5679U, last_streams_removed_.video()[0].first_ssrc());
// Duplicate removal: should be ignored.
last_streams_removed_.mutable_audio()->clear();
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamRemove("audio", "Bob", "audio1")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(0U, last_streams_removed_.audio().size());
// Duplicate removal: should be ignored.
last_streams_removed_.mutable_video()->clear();
streams_stanza.reset(buzz::XmlElement::ForStr(
JingleStreamRemove("video", "Bob", "video1")));
SetJingleSid(streams_stanza.get());
client_->session_manager()->OnIncomingMessage(streams_stanza.get());
ASSERT_EQ(0U, last_streams_removed_.video().size());
voice_channel = fme_->GetVoiceChannel(0);
ASSERT_TRUE(voice_channel != NULL);
ASSERT_FALSE(voice_channel->HasRecvStream(1234U));
ASSERT_TRUE(voice_channel->HasRecvStream(2468U));
video_channel = fme_->GetVideoChannel(0);
ASSERT_TRUE(video_channel != NULL);
ASSERT_FALSE(video_channel->HasRecvStream(5678U));
// Fails because ssrc is now invalid.
ASSERT_FALSE(
call_->SendViewRequest(call_->sessions()[0], viewRequest));
ClearStanzas();
}
void MakeSignalingSecure(cricket::SecureMediaPolicy secure) {
client_->set_secure(secure);
}
void ExpectCrypto(cricket::SecureMediaPolicy secure) {
MakeSignalingSecure(secure);
expect_incoming_crypto_ = true;
#ifdef HAVE_SRTP
expect_outgoing_crypto_ = true;
#endif
}
void ExpectVideoBandwidth(int bandwidth) {
expected_video_bandwidth_ = bandwidth;
}
void ExpectVideoRtcpMux(bool rtcp_mux) {
expected_video_rtcp_mux_ = rtcp_mux;
}
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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template <class C>
void SetCodecFeedbackParams(std::vector<C>* codecs,
const FeedbackParams& fb_params) {
for (size_t i = 0; i < codecs->size(); ++i) {
codecs->at(i).feedback_params = fb_params;
}
}
void ExpectRtcpFb() {
FeedbackParams params_nack_fir;
params_nack_fir.Add(FeedbackParam(cricket::kRtcpFbParamCcm,
cricket::kRtcpFbCcmParamFir));
params_nack_fir.Add(FeedbackParam(cricket::kRtcpFbParamNack));
FeedbackParams params_nack;
params_nack.Add(FeedbackParam(cricket::kRtcpFbParamNack));
expected_audio_fb_params_ = params_nack;
expected_video_fb_params_ = params_nack_fir;
expected_data_fb_params_ = params_nack;
}
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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private:
void OnSendStanza(cricket::SessionManager* manager,
const buzz::XmlElement* stanza) {
LOG(LS_INFO) << stanza->Str();
stanzas_.push_back(new buzz::XmlElement(*stanza));
}
void OnSessionCreate(cricket::Session* session, bool initiate) {
session->set_current_protocol(initial_protocol_);
}
void OnCallCreate(cricket::Call *call) {
call_ = call;
call->SignalMediaStreamsUpdate.connect(
this, &MediaSessionClientTest::OnMediaStreamsUpdate);
}
void OnCallDestroy(cricket::Call *call) {
call_ = NULL;
}
void OnMediaStreamsUpdate(cricket::Call *call,
cricket::Session *session,
const cricket::MediaStreams& added,
const cricket::MediaStreams& removed) {
last_streams_added_.CopyFrom(added);
last_streams_removed_.CopyFrom(removed);
}
talk_base::NetworkManager* nm_;
cricket::PortAllocator* pa_;
cricket::SessionManager* sm_;
cricket::FakeMediaEngine* fme_;
cricket::FakeDataEngine* fdme_;
cricket::MediaSessionClient* client_;
cricket::Call* call_;
std::vector<buzz::XmlElement* > stanzas_;
MediaSessionTestParser* parser_;
cricket::SignalingProtocol initial_protocol_;
bool expect_incoming_crypto_;
bool expect_outgoing_crypto_;
int expected_video_bandwidth_;
bool expected_video_rtcp_mux_;
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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FeedbackParams expected_audio_fb_params_;
FeedbackParams expected_video_fb_params_;
FeedbackParams expected_data_fb_params_;
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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cricket::MediaStreams last_streams_added_;
cricket::MediaStreams last_streams_removed_;
};
MediaSessionClientTest* GingleTest() {
return new MediaSessionClientTest(new GingleSessionTestParser(),
cricket::PROTOCOL_GINGLE);
}
MediaSessionClientTest* JingleTest() {
return new MediaSessionClientTest(new JingleSessionTestParser(),
cricket::PROTOCOL_JINGLE);
}
Update libjingle to 50654631. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
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TEST(MediaSessionTest, JingleGoodInitiateWithRtcpFb) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
cricket::CallOptions options = VideoCallOptions();
options.data_channel_type = cricket::DCT_SCTP;
test->ExpectRtcpFb();
test->TestGoodIncomingInitiate(
kJingleInitiateWithRtcpFb, options, elem.use());
}
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
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TEST(MediaSessionTest, JingleGoodVideoInitiate) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
kJingleVideoInitiate, VideoCallOptions(), elem.use());
test->TestCodecsOfVideoInitiate(elem.get());
}
TEST(MediaSessionTest, JingleGoodVideoInitiateWithBandwidth) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->ExpectVideoBandwidth(42000);
test->TestGoodIncomingInitiate(
kJingleVideoInitiateWithBandwidth, VideoCallOptions(), elem.use());
}
TEST(MediaSessionTest, JingleGoodVideoInitiateWithRtcpMux) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->ExpectVideoRtcpMux(true);
test->TestGoodIncomingInitiate(
kJingleVideoInitiateWithRtcpMux, VideoCallOptions(), elem.use());
}
TEST(MediaSessionTest, JingleGoodVideoInitiateWithRtpData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
cricket::CallOptions options = VideoCallOptions();
options.data_channel_type = cricket::DCT_RTP;
test->TestGoodIncomingInitiate(
AddEncryption(kJingleVideoInitiateWithRtpData, kJingleCryptoOffer),
options,
elem.use());
}
TEST(MediaSessionTest, JingleGoodVideoInitiateWithSctpData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
cricket::CallOptions options = VideoCallOptions();
options.data_channel_type = cricket::DCT_SCTP;
test->TestGoodIncomingInitiate(kJingleVideoInitiateWithSctpData,
options,
elem.use());
}
TEST(MediaSessionTest, JingleRejectAudio) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
cricket::CallOptions options = VideoCallOptions();
options.has_audio = false;
options.data_channel_type = cricket::DCT_RTP;
test->TestRejectOffer(kJingleVideoInitiateWithRtpData, options, elem.use());
}
TEST(MediaSessionTest, JingleRejectVideo) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
cricket::CallOptions options = AudioCallOptions();
options.data_channel_type = cricket::DCT_RTP;
test->TestRejectOffer(kJingleVideoInitiateWithRtpData, options, elem.use());
}
TEST(MediaSessionTest, JingleRejectData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestRejectOffer(
kJingleVideoInitiateWithRtpData, VideoCallOptions(), elem.use());
}
TEST(MediaSessionTest, JingleRejectVideoAndData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestRejectOffer(
kJingleVideoInitiateWithRtpData, AudioCallOptions(), elem.use());
}
TEST(MediaSessionTest, JingleGoodInitiateAllSupportedAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
kJingleInitiate, AudioCallOptions(), elem.use());
test->TestHasAllSupportedAudioCodecs(elem.get());
}
TEST(MediaSessionTest, JingleGoodInitiateDifferentPreferenceAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
kJingleInitiateDifferentPreference, AudioCallOptions(), elem.use());
test->TestHasAllSupportedAudioCodecs(elem.get());
}
TEST(MediaSessionTest, JingleGoodInitiateSomeUnsupportedAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
kJingleInitiateSomeUnsupported, AudioCallOptions(), elem.use());
test->TestHasAudioCodecsFromInitiateSomeUnsupported(elem.get());
}
TEST(MediaSessionTest, JingleGoodInitiateDynamicAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
kJingleInitiateDynamicAudioCodecs, AudioCallOptions(), elem.use());
test->TestHasAudioCodecsFromInitiateDynamicAudioCodecs(elem.get());
}
TEST(MediaSessionTest, JingleGoodInitiateStaticAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
kJingleInitiateStaticAudioCodecs, AudioCallOptions(), elem.use());
test->TestHasAudioCodecsFromInitiateStaticAudioCodecs(elem.get());
}
TEST(MediaSessionTest, JingleBadInitiateNoAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(kJingleInitiateNoAudioCodecs);
}
TEST(MediaSessionTest, JingleBadInitiateNoSupportedAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(kJingleInitiateNoSupportedAudioCodecs);
}
TEST(MediaSessionTest, JingleBadInitiateWrongClockrates) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(kJingleInitiateWrongClockrates);
}
TEST(MediaSessionTest, JingleBadInitiateWrongChannels) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(kJingleInitiateWrongChannels);
}
TEST(MediaSessionTest, JingleBadInitiateNoPayloadTypes) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(kJingleInitiateNoPayloadTypes);
}
TEST(MediaSessionTest, JingleBadInitiateDynamicWithoutNames) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(kJingleInitiateDynamicWithoutNames);
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiate) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestGoodOutgoingInitiate(AudioCallOptions());
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiateWithBandwidth) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options = VideoCallOptions();
options.video_bandwidth = 42000;
test->TestGoodOutgoingInitiate(options);
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiateWithRtcpMux) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options = VideoCallOptions();
options.rtcp_mux_enabled = true;
test->TestGoodOutgoingInitiate(options);
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiateWithRtpData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options;
options.data_channel_type = cricket::DCT_RTP;
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodOutgoingInitiate(options);
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiateWithSctpData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options;
options.data_channel_type = cricket::DCT_SCTP;
test->TestGoodOutgoingInitiate(options);
}
// Crypto related tests.
// Offer has crypto but the session is not secured, just ignore it.
TEST(MediaSessionTest, JingleInitiateWithCryptoIsIgnoredWhenNotSecured) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->TestGoodIncomingInitiate(
AddEncryption(kJingleVideoInitiate, kJingleCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has crypto required but the session is not secure, fail.
TEST(MediaSessionTest, JingleInitiateWithCryptoRequiredWhenNotSecured) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(AddEncryption(kJingleVideoInitiate,
kJingleRequiredCryptoOffer));
}
// Offer has no crypto but the session is secure required, fail.
TEST(MediaSessionTest, JingleInitiateWithNoCryptoFailsWhenSecureRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->ExpectCrypto(cricket::SEC_REQUIRED);
test->TestBadIncomingInitiate(kJingleInitiate);
}
// Offer has crypto and session is secure, expect crypto in the answer.
TEST(MediaSessionTest, JingleInitiateWithCryptoWhenSecureEnabled) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodIncomingInitiate(
AddEncryption(kJingleVideoInitiate, kJingleCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has crypto and session is secure required, expect crypto in
// the answer.
TEST(MediaSessionTest, JingleInitiateWithCryptoWhenSecureRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->ExpectCrypto(cricket::SEC_REQUIRED);
test->TestGoodIncomingInitiate(
AddEncryption(kJingleVideoInitiate, kJingleCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has unsupported crypto and session is secure, no crypto in
// the answer.
TEST(MediaSessionTest, JingleInitiateWithUnsupportedCrypto) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
talk_base::scoped_ptr<buzz::XmlElement> elem;
test->MakeSignalingSecure(cricket::SEC_ENABLED);
test->TestGoodIncomingInitiate(
AddEncryption(kJingleInitiate, kJingleUnsupportedCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has unsupported REQUIRED crypto and session is not secure, fail.
TEST(MediaSessionTest, JingleInitiateWithRequiredUnsupportedCrypto) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestBadIncomingInitiate(
AddEncryption(kJingleInitiate, kJingleRequiredUnsupportedCryptoOffer));
}
// Offer has unsupported REQUIRED crypto and session is secure, fail.
TEST(MediaSessionTest, JingleInitiateWithRequiredUnsupportedCryptoWhenSecure) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->MakeSignalingSecure(cricket::SEC_ENABLED);
test->TestBadIncomingInitiate(
AddEncryption(kJingleInitiate, kJingleRequiredUnsupportedCryptoOffer));
}
// Offer has unsupported REQUIRED crypto and session is required secure, fail.
TEST(MediaSessionTest,
JingleInitiateWithRequiredUnsupportedCryptoWhenSecureRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->MakeSignalingSecure(cricket::SEC_REQUIRED);
test->TestBadIncomingInitiate(
AddEncryption(kJingleInitiate, kJingleRequiredUnsupportedCryptoOffer));
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiateWithCrypto) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodOutgoingInitiate(AudioCallOptions());
}
TEST(MediaSessionTest, JingleGoodOutgoingInitiateWithCryptoRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->ExpectCrypto(cricket::SEC_REQUIRED);
test->TestGoodOutgoingInitiate(AudioCallOptions());
}
TEST(MediaSessionTest, JingleIncomingAcceptWithSsrcs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options = VideoCallOptions();
options.is_muc = true;
test->TestIncomingAcceptWithSsrcs(kJingleAcceptWithSsrcs, options);
}
TEST(MediaSessionTest, JingleIncomingAcceptWithRtpDataSsrcs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options = VideoCallOptions();
options.is_muc = true;
options.data_channel_type = cricket::DCT_RTP;
test->TestIncomingAcceptWithSsrcs(kJingleAcceptWithRtpDataSsrcs, options);
}
TEST(MediaSessionTest, JingleIncomingAcceptWithSctpData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
cricket::CallOptions options = VideoCallOptions();
options.is_muc = true;
options.data_channel_type = cricket::DCT_SCTP;
test->TestIncomingAcceptWithSsrcs(kJingleAcceptWithSctpData, options);
}
TEST(MediaSessionTest, JingleStreamsUpdateAndView) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestStreamsUpdateAndViewRequests();
}
TEST(MediaSessionTest, JingleSendVideoStreamUpdate) {
talk_base::scoped_ptr<MediaSessionClientTest> test(JingleTest());
test->TestSendVideoStreamUpdate();
}
// Gingle tests
TEST(MediaSessionTest, GingleGoodVideoInitiate) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleVideoInitiate, VideoCallOptions(), elem.use());
test->TestCodecsOfVideoInitiate(elem.get());
}
TEST(MediaSessionTest, GingleGoodVideoInitiateWithBandwidth) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectVideoBandwidth(42000);
test->TestGoodIncomingInitiate(
kGingleVideoInitiateWithBandwidth, VideoCallOptions(), elem.use());
}
TEST(MediaSessionTest, GingleGoodInitiateAllSupportedAudioCodecs) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleInitiate, AudioCallOptions(), elem.use());
test->TestHasAllSupportedAudioCodecs(elem.get());
}
TEST(MediaSessionTest, GingleGoodInitiateAllSupportedAudioCodecsWithCrypto) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleCryptoOffer),
AudioCallOptions(),
elem.use());
test->TestHasAllSupportedAudioCodecs(elem.get());
}
TEST(MediaSessionTest, GingleGoodInitiateDifferentPreferenceAudioCodecs) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleInitiateDifferentPreference, AudioCallOptions(), elem.use());
test->TestHasAllSupportedAudioCodecs(elem.get());
}
TEST(MediaSessionTest, GingleGoodInitiateSomeUnsupportedAudioCodecs) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleInitiateSomeUnsupported, AudioCallOptions(), elem.use());
test->TestHasAudioCodecsFromInitiateSomeUnsupported(elem.get());
}
TEST(MediaSessionTest, GingleGoodInitiateDynamicAudioCodecs) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleInitiateDynamicAudioCodecs, AudioCallOptions(), elem.use());
test->TestHasAudioCodecsFromInitiateDynamicAudioCodecs(elem.get());
}
TEST(MediaSessionTest, GingleGoodInitiateStaticAudioCodecs) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleInitiateStaticAudioCodecs, AudioCallOptions(), elem.use());
test->TestHasAudioCodecsFromInitiateStaticAudioCodecs(elem.get());
}
TEST(MediaSessionTest, GingleGoodInitiateNoAudioCodecs) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
kGingleInitiateNoAudioCodecs, AudioCallOptions(), elem.use());
test->TestHasDefaultAudioCodecs(elem.get());
}
TEST(MediaSessionTest, GingleBadInitiateNoSupportedAudioCodecs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(kGingleInitiateNoSupportedAudioCodecs);
}
TEST(MediaSessionTest, GingleBadInitiateWrongClockrates) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(kGingleInitiateWrongClockrates);
}
TEST(MediaSessionTest, GingleBadInitiateWrongChannels) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(kGingleInitiateWrongChannels);
}
TEST(MediaSessionTest, GingleBadInitiateNoPayloadTypes) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(kGingleInitiateNoPayloadTypes);
}
TEST(MediaSessionTest, GingleBadInitiateDynamicWithoutNames) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(kGingleInitiateDynamicWithoutNames);
}
TEST(MediaSessionTest, GingleGoodOutgoingInitiate) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodOutgoingInitiate(AudioCallOptions());
}
TEST(MediaSessionTest, GingleGoodOutgoingInitiateWithBandwidth) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
cricket::CallOptions options = VideoCallOptions();
options.video_bandwidth = 42000;
test->TestGoodOutgoingInitiate(options);
}
// Crypto related tests.
// Offer has crypto but the session is not secured, just ignore it.
TEST(MediaSessionTest, GingleInitiateWithCryptoIsIgnoredWhenNotSecured) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestGoodIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has crypto required but the session is not secure, fail.
TEST(MediaSessionTest, GingleInitiateWithCryptoRequiredWhenNotSecured) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(AddEncryption(kGingleInitiate,
kGingleRequiredCryptoOffer));
}
// Offer has no crypto but the session is secure required, fail.
TEST(MediaSessionTest, GingleInitiateWithNoCryptoFailsWhenSecureRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectCrypto(cricket::SEC_REQUIRED);
test->TestBadIncomingInitiate(kGingleInitiate);
}
// Offer has crypto and session is secure, expect crypto in the answer.
TEST(MediaSessionTest, GingleInitiateWithCryptoWhenSecureEnabled) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has crypto and session is secure required, expect crypto in
// the answer.
TEST(MediaSessionTest, GingleInitiateWithCryptoWhenSecureRequired) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectCrypto(cricket::SEC_REQUIRED);
test->TestGoodIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has unsupported crypto and session is secure, no crypto in
// the answer.
TEST(MediaSessionTest, GingleInitiateWithUnsupportedCrypto) {
talk_base::scoped_ptr<buzz::XmlElement> elem;
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->MakeSignalingSecure(cricket::SEC_ENABLED);
test->TestGoodIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleUnsupportedCryptoOffer),
VideoCallOptions(),
elem.use());
}
// Offer has unsupported REQUIRED crypto and session is not secure, fail.
TEST(MediaSessionTest, GingleInitiateWithRequiredUnsupportedCrypto) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->TestBadIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleRequiredUnsupportedCryptoOffer));
}
// Offer has unsupported REQUIRED crypto and session is secure, fail.
TEST(MediaSessionTest, GingleInitiateWithRequiredUnsupportedCryptoWhenSecure) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->MakeSignalingSecure(cricket::SEC_ENABLED);
test->TestBadIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleRequiredUnsupportedCryptoOffer));
}
// Offer has unsupported REQUIRED crypto and session is required secure, fail.
TEST(MediaSessionTest,
GingleInitiateWithRequiredUnsupportedCryptoWhenSecureRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->MakeSignalingSecure(cricket::SEC_REQUIRED);
test->TestBadIncomingInitiate(
AddEncryption(kGingleInitiate, kGingleRequiredUnsupportedCryptoOffer));
}
TEST(MediaSessionTest, GingleGoodOutgoingInitiateWithCrypto) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodOutgoingInitiate(AudioCallOptions());
}
TEST(MediaSessionTest, GingleGoodOutgoingInitiateWithCryptoRequired) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
test->ExpectCrypto(cricket::SEC_REQUIRED);
test->TestGoodOutgoingInitiate(AudioCallOptions());
}
TEST(MediaSessionTest, GingleIncomingAcceptWithSsrcs) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
cricket::CallOptions options = VideoCallOptions();
options.is_muc = true;
test->TestIncomingAcceptWithSsrcs(kGingleAcceptWithSsrcs, options);
}
TEST(MediaSessionTest, GingleGoodOutgoingInitiateWithRtpData) {
talk_base::scoped_ptr<MediaSessionClientTest> test(GingleTest());
cricket::CallOptions options;
options.data_channel_type = cricket::DCT_RTP;
test->ExpectCrypto(cricket::SEC_ENABLED);
test->TestGoodOutgoingInitiate(options);
}
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