[asterisk-dev] DTMF SSRC issue regression, constantssrc

[Matthew Jordan]

On Fri, Mar 7, 2014 at 12:07 PM, Daniel Pocock <daniel at pocock.com.au> wrote:
>
>
>
> Hi,
>
> I've recently observed an Asterisk 11.5 setup sending rfc2833 DTMF with
> an SSRC that is distinct from the audio stream SSRC.
>
> It was discussed in this bug which is marked "FIXED", but it is still
> happening, maybe it is a regression now:
>
> https://issues.asterisk.org/jira/browse/ASTERISK-16851

There isn't enough information here for someone to know whether or not
this is the same issue.

> Has this changed again because of the PjSIP changes?

Nope. Both chan_pjsip and chan_sip use the same media stack for RFC
2833 DTMF generation.

> Is the constantssrc option to be supported going forward?

I'm not sure what option you are referring to. There isn't a
constantssrc option in rtp.conf or in res_rtp_asterisk.

> In case it is relevant:
> - I'm using SIP
> - the DTMF comes in to Asterisk using SIP INFO over TCP

I'm assuming you have a DTMF generated by a SIP INFO request, which
creates an RFC 2833 DTMF on an outbound channel.

If that is the case, there are a number of open issues regarding SIP
INFO to RFC 2833 conversion, but most of those have to do with the
DTMF duration. You're welcome to open an issue for this, but I'd
suggest gathering all of the standard logs illustrating the problem
before doing so.

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org