2014-05-03 34 views
0

Asterisk 11无法在特定WIFI网络上传送主叫方和被叫方语音声音。Asterisk无法在特定WIFI网络上听到声音

WIFI手机==> 4G LTE手机(能听到声音/工作)

== Using SIP RTP CoS mark 5 
-- Called SIP/01036504100 
-- SIP/01036504100-00000594 is ringing 
-- SIP/01036504100-00000594 answered SIP/01010001004-00000593 
-- Locally bridging SIP/01010001004-00000593 and SIP/01036504100-00000594 
    > 0x7f5a401b6800 -- Probation passed - setting RTP source address to 1XX.63.12.134:7076 
    > 0x7f5a401b6800 -- Probation passed - setting RTP source address to 1XX.63.12.134:7076 
    > 0x7f5a3800bf90 -- Probation passed - setting RTP source address to 2XX.62.163.73:51658 

3G手机==> 4G LTE手机(能听到声音/工作)

== Using SIP RTP CoS mark 5 
-- Called SIP/01088143268 
-- SIP/01088143268-00000596 is ringing 
-- SIP/01088143268-00000596 answered SIP/01036504100-00000595 
-- Remotely bridging SIP/01036504100-00000595 and SIP/01088143268-00000596 
    > 0x7f5a3800bf90 -- Probation passed - setting RTP source address to 3X.7.29.226:2779 
    > 0x7f5a40017050 -- Probation passed - setting RTP source address to 2XX.62.163.73:51944 
    > 0x7f5a3800bf90 -- Probation passed - setting RTP source address to 3X.7.29.226:2779 

另一个WIFI电话==> 4G LTE手机(听不到声音/不工作)

== Using SIP RTP CoS mark 5 
-- Called SIP/01036504100 
-- SIP/01036504100-00000598 is ringing 
-- SIP/01036504100-00000598 answered SIP/01088143268-00000597 
-- Remotely bridging SIP/01088143268-00000597 and SIP/01036504100-00000598 
    > 0x7f5a40116470 -- Probation passed - setting RTP source address to 5X.237.58.102:7076 
    > 0x7f5a40116470 -- Probation passed - setting RTP source address to 5X.237.58.102:7076 
    > 0x7f5a38027a20 -- Probation passed - setting RTP source address to 2XX.62.163.73:52040 
    > 0x7f5a38027a20 -- Probation passed - setting RTP source address to 2XX.62.163.73:52040 

我在想也许我只在10,000到20,000之间打开UDP。但是,我错了。星号-rvvvvv不会告诉我什么是问题。

回答

1

我改变了用户的nat价值“force_rport,COMEDIA”现在无论用户可以听到声音。

nat=force_rport,comedia 

很奇怪,NAT = yes和NAT = force_rport,comdia应该是相同的,但第二个是工作Asteirks 11.

2

检查控制台上的SIP和RTP调试日志,方法是打开它们:sip set debug onrtp set debug on

通过这种方式,您可以找出RTP音频流的哪一段不会到达它应该在的位置。这主要是通过NAT问题引起的(见sip.conf的NAT部分。

,如果你不能看到从手机进入的RTP包,那么可能是防火墙阻止流量或存在NAT问题。