[Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)

[John Coll]

Thanks to Dave I now have two Grandstream phones with a voice path. Yippee!

Wanting to learn from the experience I compared the sip debugs from before
and after adding the disallow=all, allow=ulaw, allow=alaw lines to sip.conf
to see "what I should have noticed" in the debug that would have pointed me
to the problem.

I see that during negotiation I got the following

Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0

Now why didn't asterisk and grandstream negotiate a common codec?
Why does only allowing ulaw and alaw work better than allowing everything?

That sounds to me as if one or other end is failing to negotiate correctly.

Interestingly if I remove both of the allow lines for both phones I get
*CLI> WARNING[81926]: File chan_sip.c, Line 1954 (process_sdp): No
compatible codecs!
but the voice path works fine - perhaps using no compression???


In a FAQ I read:

Q What Codec should I use for my Granstream phone?
(http://www.grandstream.com/FAQ.htm#Q15)

A: By default, PCMU(G711u) will be used. Both PCMU and PCMA will give you
toll quality but their bandwidth consumption is also the highest (64kbps).
If your network bandwidth is low, you can choose other lower-bit-rate codec
such as G723 or G729 which will give you near toll quality at much smaller
bandwidth consumption (G723 consumes 5.3/6.3kbps and G729 consumes 8kbps).
If bandwidth is not a concern and you want good voice quality, try using
PCMU or PCMA, or even the new wide band codec G722 (64kbps) which will
provide hi-fidelity voice that is better than toll quality.


The phrase "by default" seems to imply that negotiation will sort out the
codecs.

Eduardo Goncalves seems to raise essentially the same issue on December 16,
2003, "Re: [Asterisk-Users] codec negotiation" but no answer seems to
emerge.

I've pretty much had enough of this problem so don't spend long on a
detailed response but I am curious to know why * and the GS phones failed to
negotiate the right codec. Is there a bug / "incompatibility" issue?

thanks
john




-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Dave Cotton
Sent: 03 January 2004 18:11
To: Asterisk List
Subject: RE: [Asterisk-Users] Newbie - getting two local
phonestocommunicate would be a good start :)


On Sat, 2004-01-03 at 18:59, John Coll wrote:
> Dave
>
> You were right!
>
>
In the words of that welsh comedian "I know because I was there".

As others have said there's a steep learning curve for *, but as one
who's climbed just some of it, I can say it's worth it.


--
Dave Cotton <dcotton at linuxautrement.com>

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