Asterisk cannot hear sound on specific WIFI network

StackOverflow https://stackoverflow.com/questions/23443211

  •  14-07-2023
  •  | 
  •  

Question

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.

Was it helpful?

Solution 2

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.

OTHER TIPS

Examine SIP and RTP debug logs on the console by turning them on: sip set debug on, and rtp set debug on.

This way you could find out which leg of the RTP audio stream is not going to where it should. This is caused mainly by NAT issues (see the NAT section of sip.conf.

If You can't see incoming RTP packets from the phone, then probably a firewall is blocking the traffic or there is a NAT issue.

Licensed under: CC-BY-SA with attribution
Not affiliated with StackOverflow
scroll top