[Asterisk-Users] RTP timing in a SIP only world (choppy MOH)

[Bob Knight]

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?

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