2014-05-03 51 views
0

Asterisk 11無法在特定WIFI網絡上傳送主叫方和被叫方語音聲音。Asterisk無法在特定WIFI網絡上聽到聲音

WIFI手機==> 4G LTE手機(能聽到聲音/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手機==> 4G LTE手機(能聽到聲音/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 

另一個WIFI電話==> 4G LTE手機(聽不到聲音/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 

我在想也許我只在10,000到20,000之間打開UDP。但是,我錯了。星號-rvvvvv不會告訴我什麼是問題。

回答

1

我改變了用戶的nat價值「force_rport,COMEDIA」現在無論用戶可以聽到聲音。

nat=force_rport,comedia 

很奇怪,NAT = yes和NAT = force_rport,comdia應該是相同的,但第二個是工作Asteirks 11.

2

檢查控制檯上的SIP和RTP調試日誌,方法是打開它們:sip set debug onrtp set debug on

通過這種方式,您可以找出RTP音頻流的哪一段不會到達它應該在的位置。這主要是通過NAT問題引起的(見sip.conf的NAT部分。

,如果你不能看到從手機進入的RTP包,那麼可能是防火牆阻止流量或存在NAT問題。

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