Stanza.io 无法创建 WebRTC 答案

Question

我正在使用 ejabberd + stanza.io 制作一个实时消息传递网站，一切正常。现在我愿意使用 jingle 协议来实现 Webrtc audio/video。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();

问题是当我从其他应用程序（AstraChat：iOS 支持 Jingle 的应用程序）呼叫用户时，我在浏览器日志中得到这个：

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，我们将不胜感激。

Answer 1

根据您的日志，来自 Astra 聊天的会话启动是这样的： <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> 它只提供 WebRTC 不支持的 speex 音频编解码器（opus 或 G.711 是强制性的）。此外，WebRTC 中也没有强制性的加密。

Stanza.io 无法创建 WebRTC 答案

Stanza.io Could not create WebRTC answer

javascript

node.js

xmpp

ejabberd

stanza.io