Google chrome sipml5星号11.7单向音频

我已将sipml5与我的asterisk服务器（可公开访问）集成。我面临两个sipxml5客户端之间的单向音频通话

补充资料

1.如果一个客户端为xlite，另一个客户端为sipml5，则工作正常

2.ICE在sipml5客户端和asterisk中均已启用

3.两个客户端位于chrome的不同选项卡中

4.1我的sip配置

transport=udp,ws,wss
qualify=yes
encryption = yes
avpf = yes
directmedia = outgoing
allow=ulaw
type=friend
dtmfmode=rfc2833
insecure=invite,port
qualify=yes
host=dynamic
call-limit=1
alwaysauthreject=yes
nat=force_rport,comedia

5.这是我的rtp登录星号

Sent RTP packet to      xxx.xx.xxx.x:57214 (via ICE) (type 00, seq 048742, ts 397541448, len 4294967284)
Got  RTP packet from    xxx.xx.xxx.x:57214 (type 00, seq 013761, ts 397541345, len 000160)

Sent RTP packet to      xxx.xx.xxx.x:57210 (type 00, seq 040424, ts 397541344, len 000164)
Got  RTP packet from    xxx.xx.xxx.x:57210 (type 00, seq 031460, ts 397541609, len 000160)

Sent RTP packet to      xxx.xx.xxx.x:57214 (via ICE) (type 00, seq 048743, ts 397541608, len 4294967284)
Got  RTP packet from    xxx.xx.xxx.x:57214 (type 00, seq 013762, ts 397541505, len 000160)

Sent RTP packet to      xxx.xx.xxx.x:57210 (type 00, seq 040425, ts 397541504, len 000164)
Got  RTP packet from    xxx.xx.xxx.x:57210 (type 00, seq 031461, ts 397541769, len 000160)

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Sipml是非常专业的

它如何工作在很大程度上取决于客户端的重新大小和使用的星号版本（甚至是subversion！！！）

因此，对于任何使用sipml的情况，都需要专家级调试

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如果它与sipml->sip完美配合，请尝试调用sipml->Local/s@context/n->sipml。这将强制执行编解码器转换，并通过星号发送所有rtp。

是否已在星号中启用videosupport=yes？？如果在asterisk中启用视频，则有时会发生这种情况，因为asterisk会向被叫方发送视频SDP