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Asterisk在特定的WIFI网络上听不到声音

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Asterisk 11无法在特定的WIFI网络上传送来电和被叫语音 .

WIFI phone ==> 4G LTE phone (Can hear sound/Working)

== 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 phone ==> 4G LTE phone (Can hear sound/Working)

== 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

Another WIFI phone ==> 4G LTE phone (Can't hear sound/Not Working)

== 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

我想也许我只打开10,000到20,000之间的UDP . 但是，我错了 . asterisk -rvvvvv没有告诉我是什么问题 .

wifi sip asterisk 3g 4g

2 回答

  • 1

    通过打开控制台上的SIP和RTP调试日志来检查它们： sip set debug on 和 rtp 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应有的位置 . 这主要是由NAT问题引起的（参见sip.conf的NAT部分 .

    如果您无法看到来自手机的传入RTP数据包，则可能是防火墙阻止了流量或存在NAT问题 .

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  • 2

    我将用户的nat值更改为“force_rport，comedia”，现在两个用户都可以听到声音 .

    nat=force_rport,comedia

    奇怪的是，nat = yes和nat = force_rport，comdia应该是相同的，但第二个正在研究Asteirks 11 .

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