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

[David Nembrot]

   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.

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   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.

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