我正在使用ejabberd + stanza.io来制作一个实时消息网站,一切都运行得很完美。现在,我愿意使用jingle协议实现Webrtc音频/视频。这是我用来连接的JS代码:
var client = XMPP.createClient({
jid: xmpp_user+'@'+XMPP_DOMAIN,
password: cu.auth.user_password,
transport: 'websocket',
wsURL: "ws://"+xms+":5280/websocket/"
});
client.jingle.config.debug = true;
client.on('session:started', function () {
client.enableCarbons();
client.getRoster(function (err, resp) {
client.updateCaps();
client.sendPresence({
caps: client.disco.caps
});
});
});
client.connect();问题是,当我从另一个应用程序(AstraChat:支持静乐的iOS应用程序)呼叫用户时,我在浏览器日志中得到以下内容:
Jingle: 67bzrsog243: session-initiate undefined undefined
Jingle: 67bzrsog243: Could not create WebRTC answer undefined undefined这是astrachat发送的xml:
<iq xmlns='jabber:client' xml:lang='en' to='c4ca4238a0b923820dcc509a6f75849b@h2745110.stratoserver.net/352555070032013318140962' from='med@h2745110.stratoserver.net/AstraChat-iOS-21820150' type='get' id='3e8kjajc22'><query xmlns='http://jabber.org/protocol/disco#info'/></iq>
<r xmlns='urn:xmpp:sm:3'/>
<a h='1' xmlns='urn:xmpp:sm:3'/>
<a h='2' xmlns='urn:xmpp:sm:3'/>
jingle:created
iq:set:jingle
<iq xmlns='jabber:client' xml:lang='en' to='c4ca4238a0b923820dcc509a6f75849b@h2745110.stratoserver.net/352555070032013318140962' from='med@h2745110.stratoserver.net/AstraChat-iOS-21820150' type='set' id='3e8kjajc23'><jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' initiator='med@h2745110.stratoserver.net/AstraChat-iOS-21820150' responder='c4ca4238a0b923820dcc509a6f75849b@h2745110.stratoserver.net/352555070032013318140962' sid='3e8kjajc24'><content creator='initiator' name='voice'><description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'><payload-type id='101' name='speex' clockrate='8000'/></description><transport xmlns='urn:xmpp:jingle:transports:ice-udp:1' pwd='TC5NsD6IEQGXeDO8d5/3OU' ufrag='yA0z'/></content></jingle></iq>
<r xmlns='urn:xmpp:sm:3'/>
Jingle: 3e8kjajc24: session-initiate undefined undefined
Could not create WebRTC answer undefined undefined我真的不知道如何解决这个问题,任何帮助都将不胜感激。
发布于 2017-12-09 20:01:06
根据你的日志,来自Astra chat的session-initiate是这样的:<iq xmlns="jabber:client" xml:lang="en" to="c4ca4238a0b923820dcc509a6f75849b@h2745110.stratoserver.net/352555070032013318140962" from="med@h2745110.stratoserver.net/AstraChat-iOS-21820150" type="set" id="3e8kjajc23"> <jingle xmlns="urn:xmpp:jingle:1" action="session-initiate" initiator="med@h2745110.stratoserver.net/AstraChat-iOS-21820150" responder="c4ca4238a0b923820dcc509a6f75849b@h2745110.stratoserver.net/352555070032013318140962" sid="3e8kjajc24"> <content creator="initiator" name="voice"> <description xmlns="urn:xmpp:jingle:apps:rtp:1" media="audio"> <payload-type id="101" name="speex" clockrate="8000" /> </description> <transport xmlns="urn:xmpp:jingle:transports:ice-udp:1" pwd="TC5NsD6IEQGXeDO8d5/3OU" ufrag="yA0z" /> </content> </jingle> </iq>只提供WebRTC不支持的speex音频编解码器(opus或G.711是强制性的)。此外,在WebRTC中也没有强制性的加密。
https://stackoverflow.com/questions/47727407
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