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.