jacklopessapo / support-tools

Automatically exported from code.google.com/p/support-tools
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When OPUS used only one direction audio has DynamicRTP #133

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
I've made a capture of call using OPUS codec.
From CSipSimple I see
Media Description, name and address (m): audio 16384 RTP/AVP 111 101
The same string was from server.
The call has payloads like the following
CSipsimple --> server Payload type: DynamicRTP-Type-111 (111)
server --> CSipsimple Payload type: opus (111)
So, they were different

At the sametime I tried to use another brand SIP app with same server and codec 
pack there. Here are next trace from app
Media Description, name and address (m): audio 5004 RTP/AVP 123 101
The samewas from server. And next

App --> server  Payload type: DynamicRTP-Type-123 (123)
server --> app  Payload type: DynamicRTP-Type-123 (123)
Now types are the same and both are dynamic.

Server asterisk has opus pack 1.0.3-1.el5

Please check this issue to use 123 codec

With best regards

Original issue reported on code.google.com by sab....@gmail.com on 11 Sep 2015 at 9:16

GoogleCodeExporter commented 9 years ago

Original comment by chrsm...@google.com on 11 Sep 2015 at 9:19