[asterisk-users] Codec negotiation

[Mindaugas Kezys]

Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch

Regards,
Mindaugas Kezys

Kolmisoft UAB 
VoIP Billing Solutions
e-mail: info at kolmisoft.com
URL: http://www.kolmisoft.com


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Davies
Sent: Tuesday, June 29, 2010 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec negotiation

On 26 June 2010 22:08, Ryan Wagoner <rswagoner at gmail.com> wrote:
> I have Polycom phones that support the g722 codec. Adding allow=g722
> to the [general] section of sip.conf works great and I can make calls
> between the phones using g722. However Asterisk is negotiating g722
> for calls going out my voip provider and transcoding these to ulaw. In
> sip.conf for the provider I have deny=all and allow=ulaw. This can
> cause potential audio degrading and wastes cpu cycles. If Asterisk
> knows the trunk only supports ulaw why would it offer g722 to the
> phone.
>
> Ryan

Because the codec is already chosen before the call is made, and you
told it that g722 is permitted.

There are all sorts of discussions in play about codec negotiation,
but at this point in time, if you want different behaviour you'll need
to go and code it yourself, and cross-channeltype this is not going to
be trivial :)

Cheers,
Steve

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