asterisk users - Nov 2003 - 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
bk@minusw.com
925-449-9163

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.
-- 
Juanjo sin .sig

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@minusw.com
925-449-9163

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