stanza.io无法创建WEBRTC答案

Question

im使用ejabberd stanza.io创建实时消息网站，一切正常。现在，我愿意使用Jingle Protocole实施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：Support os Applachat in Suppand jingle）调用用户时，我会在浏览器日志中获取此内容：

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='[e-mail]/352555070032013318140962' from='[e-mail]/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='[e-mail]/352555070032013318140962' from='[e-mail]/AstraChat-iOS-21820150' type='set' id='3e8kjajc23'><jingle xmlns='urn:xmpp:jingle:1' action='session-initiate' initiator='[e-mail]/AstraChat-iOS-21820150' responder='[e-mail]/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，任何帮助将不胜感激。

Answer 1

根据您的日志： <iq xmlns="jabber:client" xml:lang="en" to="[e-mail]/352555070032013318140962" from="[e-mail]/AstraChat-iOS-21820150" type="set" id="3e8kjajc23"> <jingle xmlns="urn:xmpp:jingle:1" action="session-initiate" initiator="[e-mail]/AstraChat-iOS-21820150" responder="[e-mail]/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> 仅提供WEBRTC不支持的Speex Audio编解码器（Opus或G.711是强制性的）。此外，WebRTC中没有强制性的加密。