[Asterisk-Users] Silence suppression on SIP calls generated from Asterisk?

[Brian Cuthie]

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Olle E. Johansson
> 
> Brian Cuthie wrote:
> > 
> > Let's say that I have a call coming in to Asterisk through 
> a TDM400P 
> > and going out through SIP to someone on the Internet. Is there any 
> > configuration option that would allow me to do silence 
> suppression on 
> > the RTP stream generated by Asterisk on behalf of the TDM400P 
> > connected user?  SIP phones allow me to do this easily, but 
> I'd like 
> > to be able to conserve upstream bandwidth on calls that 
> don't emanate 
> > from a SIP phone here at my location.
> Asterisk SIP does not support silence suppression. In fact, 
> using Silence suppression on an inbound RTP stream will lead 
> to problems, since Asterisk takes timing from inbound RTP streams.
> 

Yeah, funny thing is I saw this problem just last night while messing around
with music on hold. I had VAD on the SIP phone on and the MOH would stop
unless I talked. I thought it was quite weird when it happened; now it makes
sense. 

I've heard that Asterisk derives its timing in strange ways, but I've been
wondering why it doesn't use the machine's clock (real-time interrupt, not
wall-clock).

-brian