One way audio in webrtc configured asterisk server

Asterisk Asterisk WebRTC
1

Hai guys,

I have successfully enabled webrtc configurations in asterisk 16. I can able make a outbound voice calls. But audio is not transmitting. I have recorded audio from callee in asterisk server. Audio is coming from caller but not able to hear the same by caller.

   > 0x7efc80011040 -- Strict RTP learning after ICE completion
   > 0x7efc7802c6c0 -- Strict RTP learning after ICE completion
   > 0x7efc7802c6c0 -- Strict RTP switching to RTP target address x.x.x.x:30275 as source
   > 0x7efc80011040 -- Strict RTP switching to RTP target address x.x.x.x:33201 as source
   > 0x7efc80011040 -- Strict RTP learning complete - Locking on source address x.x.x.x:33201
   > 0x7efc7802c6c0 -- Strict RTP learning complete - Locking on source address x.x.x.x:30275

Can anyone please help to resolve this one

2

I think a lot more information is needed to help you solve this problem.

Can you please post more log messages ?

3

As well please enable SIP tracing in the log and check the negotiated audio ports (a pcap would suffice as well). Then enable rtp debug in Asterisk:

*CLI> rtp set debug on

And compare the ports for incoming, and outgoing audio to make sure audio is flowing to/from the expected ports using the negotiated codecs.

Note you can turn rtp debugging off using the following:

*CLI> rtp set debug off
4

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