[asterisk-users] SIREN14 call setup and record/playback

[Kevin P. Fleming]

Tom Browning wrote:
> 
> I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of
> Asterisk and I'm trying to get it to accept a SIREN14 call from
> Polycom's softphone.  Having trouble with SDP negotiation, I want to
> only allow SIREN14 and nothing else.  I also want to record and playback
> files, any tips on what the Record function parameters should be?

First, don't enable any codecs labeled 'SIREN7' or 'SIREN14' on the
Polycom phone; those are pre-standard names, and they work slightly
differently than the ITU standardized codecs.

> In sip.conf I have:
> 
> disallow=all                   ; First disallow all codecs
> allow=siren14                ;  Is this the right name?

Yes, this is correct. Asterisk would also accept 'g.7221c', which is the
ITU standardized name.

> And the INVITE comes from the Polycom softphone with an SDP of:
> 
> ...
> User-Agent: Polycom VV 8.0.4.4035.
> ...
> m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8.
> a=rtpmap:99 SIREN14/16000.
> a=fmtp:99 bitrate=48000.
> a=rtpmap:98 SIREN14/16000.
> a=fmtp:98 bitrate=32000.
> a=rtpmap:97 SIREN14/16000.
> a=fmtp:97 bitrate=24000.

These three are all actually Siren7 (16kHz sample rate), but the phone
is offering them three times... I've just emailed Polycom about another
one of their phones doing this as well. These should all go away if you
disable 'Siren14' in the Polycom phone configuration.

> a=rtpmap:102 G7221/16000.
> a=fmtp:102 bitrate=32000.
> a=rtpmap:101 G7221/16000.
> a=fmtp:101 bitrate=24000.
> a=rtpmap:103 G7221/16000.
> a=fmtp:103 bitrate=16000.

These are also Siren7 (G.722.1, 16kHz sample rate), at various bit
rates. Asterisk supports the 32kbps bit rate, so if you had
'allow=g.7221' or 'allow=siren7' in sip.conf, the first of these options
would be accepted.

What you want to see in the SDP for ITU G.722.1C is 'G7221/32000' with a
bitrate of 48000. If you can find a phone configuration that results
that SDP offer, you'll be good to go.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org