Re: [Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
Hi! 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? Yep, unfortunately. That's why for example in X-Lite you'll need to change settings to Transmit Silence=Yes. No clue how to do that on the GS, I don't own any of these. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
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 [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
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? Its 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. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
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? Its 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 [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
