wifisipasterisk3g4g

Asterisk cannot hear sound on specific WIFI network

Asterisk 11 cannot deliver caller and callee voice sound on specific WIFI network.

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

I was thinking maybe I only open UDP between 10,000 and 20,000. However, I was wrong. asterisk -rvvvvv doesn't show me what is the problem.

Solution

  • I changed user's nat value to "force_rport,comedia" and now both users can hear voice.

    nat=force_rport,comedia

    It was strange, nat = yes and nat = force_rport,comdia should be same, but second one was working on Asteirks 11.