[Asterisk-Dev] SIP and IAX: Changing codec mid-call (NOT forcing codec at dial)

[Daniel Floyd]

Hello all:

I've spent the last week looking into this trying to see if there is
any support for it and I can't find any.  There appears to be ways to
override/force a codec at dial time, at least for SIP.  IAX appears to
have something in CVS HEAD, but I can't find any docs/explanation for
it and haven't dug into the code to figure it out yet.  But either
way, that's not what I'm looking for.

I want a call (specifically incoming) to start out as ULAW (via
allow/disallow statements in sip.conf and iax.conf) so I can do fax
detection on it (although G726 seems to be pretty reliable for fax in
my testing as well).  If the call is a fax, leave the call unchanged.
If the call isn't a fax, I want to change the codec to GSM.  I know
both SIP and IAX protocols support this -- SIP via re-INVITE and IAX
by sending a new full VOICE packet with the new codec.  Although, I'm
not clear in IAX if that actually makes the far end change its codec
as well, which would be desired in this situation.

What would be great is a dialplan app that can be called to do this
dirty work.  For example:

In extensions.conf:

[incoming]
exten => s,1,Answer
exten => s,2,Ringing
exten => s,3,NVFaxDetect(|td)           ;Great App, by the way!
exten => s,4,ChangeCodec(gsm)
exten => s,105,NoOp(Codec changed failed!)  ;Do something here if
codec change fails.
exten => s,106,Hangup
exten => s,5,Dial(SIP/${OPERATOR})

exten => fax,1,Dial(SIP/${FAX})

I realize this is perhaps a little oversimpliflied since changing
codecs is channel specific, so maybe SIPChangeCodec or IAXChangeCodec
would probably be more realistic, but this gives the basic idea.

Outbound calls are a easier since I know where the call is coming
from.  But I'm running CVS stable and right now the only way I see to
do this with IAX is to define multiple IAX peers for the same host
with the only difference being the allowed codec.  Seems a little
cluggy.  Maybe the new stuff I saw in CVS HEAD gets around this, but
like I said, I couldn't find anything about it with Google or on
voip-info.org.

Any thoughts?

And I'm not opposed to trying my hand at this myself.  I do pose as a
software engineer by day.  And I could probably even make this into
work-related work.  But any pointers from anyone familiar with the IAX
and SIP code would be great as far as how simple or complex this might
be based on where the code is at right now.  If it would require a lot
of "pot stirring", I don't know that I would want to mess with it, at
least right now.  And at the very least, I wanted to generate some
discussion on the topic since I couldn't seem to find anything else on
it.

Thanks,
Daniel