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brian at freeswitch.org
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 Posted: Mon Aug 10, 2009 12:07 pm Post subject: [Freeswitch-users] Freeswitch :: VoIP issues using ILBC audi |
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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:
| Quote: | 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.
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david.nembrot at soget...
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| Posted: Mon Aug 10, 2009 12:14 pm Post subject: [Freeswitch-users] Freeswitch :: VoIP issues using ILBC audi |
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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 & 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. |
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