[Freeswitch-users] No audio on caller side when both side support speex/8000 only

[Brian West]

This looks and sounds like a case where pjsip isn't listening to our  
SDP.  If we 200 OK with speex on 102 and the far end starts sending it  
on 98 then I suspect the client is broken if I'm not mistaken.

/b

On Sep 9, 2009, at 6:19 AM, Tzury Bar Yochay wrote:

> Hi,
>
> Owe to the network bandwidth limitations (running on cellular phones
> ip link) we are using speex/8000 as our voice codec.
>
> However, when both parties are using that codec the sound is not to be
> heard on the caller side.
>
> looking at the log dumps one can see that
>
> a) at the caller side, it supports speex/8000 in pt=102 and receives
> from the server speex/8000 in pt=102
> b) at the callee side FreeSwitch supports support speex/8000 in pt=98
> although it receives from the client speex/8000 in pt=102
>
> When the voice starts caller sends RTP with pt=102 and expect to
> receive RTP with pt=102, while the callee sends RTP with pt=98 and
> expect to receive RTP with pt=102.
>
> The RTP packets that received in the caller side are with pt=98
> instead of 102 and thusly the client drops them.
>
> Attached are the 2 files recorded from a call between 2 pjsip clients
> that support only speex/8000 codec.
>
> un_FSCallerSide-speexClient.TXT – is the caller side SIP messages.
>
> un_FSAnswerSide-speexClient.TXT – is the answer side of SIP messages.
>
>
> Is there anything can be done at the configuration level to avoid  
> this?
> Thanks in advance for your help
>
> /tzury
> <un_FSAnswerSide-speexClient.TXT><un_FSCallerSide- 
> speexClient.TXT>_______________________________________________
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