[Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
Bob Knight
bk at minusw.com
Wed Nov 19 14:55:29 MST 2003
Juan, thank you very much.
Turning off VAD did it.
All is well.
Juan J. Sierralta P. wrote:
>On Wed, 2003-11-19 at 16:10, Bob Knight wrote:
>
>
>>I have an * setup with sip phones and sip fxo gateway.
>>When a sip phone places a sip/fxo call on hold, MOH is very choppy.
>>
>>It looks like RTP has a real problem with timing if it is not receiving
>>RTP packets. If the outside call that is placed on hold is not generating
>>any audio, the sip/fxo gateway does not send * RTP packets.
>>Is this valid?
>>Is this a problem with the sip/fxo gateway or a problem with * RTP timing?
>>
>>
>
> It´s known problem, Asterisk SIP channels get the timing from the
>source, so if the source stops transmitting (i.e. VAD) the MoH gets
>choppy. Try disabling VAD on your Media Gateway.
> When VAD is active it is usually signaled by an specific RTP payload
>type, maybe the SIP channel should check that an starts using a local
>clock.
>
>
>
>>Sip phone to sip phone works fine.
>>I connect 2 GS and place one on hold.
>>The GS that is receiving MOH from * is working great because the GS
>>keeps sending back RTP packets.
>>
>>IAX connections work fine.
>>I call an extension on another * box and place it on hold.
>>MOH over IAX/IAX2 is great.
>>
>>
--
Bob Knight
[-w] the work option
bk at minusw.com
925-449-9163
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