[Freeswitch-users] Freeswitch :: VoIP issues using ILBC audio codec

[Brian West]

Well since the codec is in the dynamic range the payload number  
doesn't matter.  I would have to see an RTP trace to see this... you  
must also include all the SIP traffic so wireshark can properly figure  
things out.

/b

On Aug 10, 2009, at 11:56 AM, David Nembrot wrote:

> Hello world,
>
> I've got two FS servers configured as gateways for each other and  
> I'm currently testing the telephony. Usinge the ILBC audio codec, I  
> figured out that one of the FS servers doesn't forward RTP streams  
> correctly to the other server. Here is its status-quo:
> INPUT = proper ILBC payload type (97 or 108)
> OUTPUT = unknown payload type (97 or 102)
>
> I've already changed the parameters in internal.xml & external.xml:
> <param name="inbound-codec-negotiation" value="greedy"/>
> <param name="disable-transcoding" value="true"/>
>
> When dialing out, I also use the following syntax: 
> {absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber
>
> Is there another thing to do to have proper ILBC streams passing  
> through the gateways ?
> Thanking y'all in advance ;)
>
> BR,
> David N.
>
> Hello world,
>
> I've got two FS servers configured as gateways for each other and  
> I'm currently testing the telephony. Usinge the ILBC audio codec, I  
> figured out that one of the FS servers doesn't forward RTP streams  
> correctly to the other server. Here is its status-quo:
>
> INPUT = proper ILBC payload type (97 or 108)
> OUTPUT = unknown payload type (97 or 102)
>
> I've already changed the parameters in internal.xml and also in  
> external.xml:
> <param name="inbound-codec-negotiation" value="greedy"/>
> <param name="disable-transcoding" value="true"/>
>
> Is there another thing to do to have proper ILBC streams passing  
> through the gateways ?
> Thanking y'all in advance ;)
>
> BR,
> David N.
>