[asterisk-dev] RFC 3264 - RTP media stream should be set to recvonly when recording messages via a SIP channel

[Iain McBride]

Hi Russell,

On Fri, 1 Feb 2008, Russell Bryant wrote:

> > The root cause is a particular behavior of this PSTN gateway.  When no 
> > media (RTP) packets are seen in one direction or the other for 30 seconds, 
> > the PSTN gateway assumes that the call has died and tears it down.  This 
> > seems to be an interpretation of section 5.1 of RFC 3264, which says that 
> > the direction attributes of the media stream MUST be set appropriately.
> 
> Thanks for the well written message.  :)

Thanks for the helpful response to what probably appears to be a rookie 
mistake.

> There is an option that should be a quick fix for you.  Set the
> "transmit_silence_during_record" option to "yes" in the [options] section of
> asterisk.conf.

Figures! When I query Google with that option name specifically, I find 
other users with the same issue.  Hopefully this thread will be indexed so 
that those experiencing the "30 second voicemail" issue will find this 
workaround more quickly.  

While it's more specific to the issue than the shotgun of using an 
rtpkeepalive, it seems to me that sending silence in one direction is a 
workaround for the root cause of the issue.  What are your thoughts on 
adding a method to set the RTP stream attributes appropriately when we 
intend to just receive (or send) audio?

As we move away from PSTN gateways and have more SIP-SIP end-to-end calls, 
I feel that we're going to need to be prepared to follow the RFC-specified 
behaviour.

Regards,

---
Iain McBride
Convoke Communications Corp.
http://www.convoke.com/

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