[Asterisk-Users] FireFly and GS-BT100 codec negotiation problem

[Willem de Groot]

Summary: how to force the alaw codec upon a call between Firefly & 
Grandstream BT100?

Not sure whether this is a problem with FireFly, with Asterisk, with 
both or just with me ;-)

I have:
    disallow=all
    allow=alaw
in the general section of my sip.conf.

Using Ethereal on the PC running FireFly, I get the following results.

GS = grandstream BT100 initiating SIP call, FF = firefly receiving call

* -> FF: invite from GS to FF (PCMA)
FF -> *: OK (PCMA)
* -> FF: reinvite to GS IP (PCMA, G723, PCMU, G726-32, G729, iLBC)
FF -> *: OK (iLBC)
GS -> FF: RTP stream (unanswered by FF)
[...RTP monologue continues...]
[...hangup on the GS...]
GS -> FF: RTCP goodbye
* -> FF: reinvite to * ip (PCMA)
FF -> *: OK (PCMA)
* -> FF: BYE
FF -> *: OK

Meanwhile on the *-console:
    -- Attempting native bridge of SIP/grandstream-0f65 and SIP/willem-2cbe
Oct 18 15:15:39 WARNING[98311]: chan_sip.c:2804 process_sdp: No 
compatible codecs!
Oct 18 15:15:39 WARNING[98311]: chan_sip.c:2804 process_sdp: No 
compatible codecs!

Meanwhile on the FF PC with DebugView running:
[368] Wrong input size for iLBC - requires 30ms frames
[368] Stopping transmission due to send error
[368] Finished reading

Apparently FF does not accept 20ms iLBC frames (which is the default on 
the GS phones).

So my questions:

    * Why is * advertising codecs in the reinvite request which
      shouldn't be used according to sip.conf? It advertises fine in the
      initial invite.
    * Why does FF answer "OK (iLBC)" upon the reinvite request, even
      though I have turned off the iLBC codec in FF's configuration?
      Probably a bug with FF?
    * Why does * croak "no compatible codecs" when in fact both sides
      have pcma/alaw enabled and even advertise them?

Thanks!

Willem