No subject
Sun Jul 19 19:54:31 CDT 2009
have problems with outgoing calls. When I tried this, the same way you did,
I could make calles externally but had no audio each way reguardless of what
I tried to pass to the sip provider. Best bet is to use what your sip
provider can use or find another provider that that can do g722. That's what
I did when I wanted to use g726.
my2cents
On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys <mkezys at gmail.com> wrote:
> 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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
> --
> _____________________________________________________________________
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<div>From what I have seen if your sip provider does not=A0take g722 then y=
ou will have problems with outgoing calls. When I tried this, the same way =
you did, I could make calles externally but had no audio each way reguardle=
ss of what I tried to pass to the sip provider. Best bet is to use what you=
r sip provider can use or find another provider that that can do g722. That=
's what I did when I wanted to use g726.</div>
<div>=A0</div>
<div>my2cents<br><br></div>
<div class=3D"gmail_quote">On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys=
<span dir=3D"ltr"><<a href=3D"mailto:mkezys at gmail.com">mkezys at gmail.com=
</a>></span> wrote:<br>
<blockquote style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex=
; PADDING-LEFT: 1ex" class=3D"gmail_quote">Try this: <a href=3D"http://www.=
b2bua.org/wiki/AsteriskCodecNegotiationPatch" target=3D"_blank">http://www.=
b2bua.org/wiki/AsteriskCodecNegotiationPatch</a><br>
<br>Regards,<br>Mindaugas Kezys<br><br>Kolmisoft UAB<br>VoIP Billing Soluti=
ons<br>e-mail: <a href=3D"mailto:info at kolmisoft.com">info at kolmisoft.com</a>=
<br>URL: <a href=3D"http://www.kolmisoft.com/" target=3D"_blank">http://www=
.kolmisoft.com</a><br>
<div>
<div></div>
<div class=3D"h5"><br><br>-----Original Message-----<br>From: <a href=3D"ma=
ilto:asterisk-users-bounces at lists.digium.com">asterisk-users-bounces at lists.=
digium.com</a><br>[mailto:<a href=3D"mailto:asterisk-users-bounces at lists.di=
gium.com">asterisk-users-bounces at lists.digium.com</a>] On Behalf Of Steve D=
avies<br>
Sent: Tuesday, June 29, 2010 7:51 PM<br>To: Asterisk Users Mailing List - N=
on-Commercial Discussion<br>Subject: Re: [asterisk-users] Codec negotiation=
<br><br>On 26 June 2010 22:08, Ryan Wagoner <<a href=3D"mailto:rswagoner=
@gmail.com">rswagoner at gmail.com</a>> wrote:<br>
> I have Polycom phones that support the g722 codec. Adding allow=3Dg722=
<br>> to the [general] section of sip.conf works great and I can make ca=
lls<br>> between the phones using g722. However Asterisk is negotiating =
g722<br>
> for calls going out my voip provider and transcoding these to ulaw. In=
<br>> sip.conf for the provider I have deny=3Dall and allow=3Dulaw. This=
can<br>> cause potential audio degrading and wastes cpu cycles. If Aste=
risk<br>
> knows the trunk only supports ulaw why would it offer g722 to the<br>&=
gt; phone.<br>><br>> Ryan<br><br>Because the codec is already chosen =
before the call is made, and you<br>told it that g722 is permitted.<br>
<br>There are all sorts of discussions in play about codec negotiation,<br>=
but at this point in time, if you want different behaviour you'll need<=
br>to go and code it yourself, and cross-channeltype this is not going to<b=
r>
be trivial :)<br><br>Cheers,<br>Steve<br><br>--<br>________________________=
_____________________________________________<br>-- Bandwidth and Colocatio=
n Provided by <a href=3D"http://www.api-digital.com/" target=3D"_blank">htt=
p://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>=
=A0 =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.asterisk.org/hello" targe=
t=3D"_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailin=
g list<br>To UNSUBSCRIBE or update options visit:<br>
=A0 <a href=3D"http://lists.digium.com/mailman/listinfo/asterisk-users" tar=
get=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><=
br><br><br>--<br>__________________________________________________________=
___________<br>
-- Bandwidth and Colocation Provided by <a href=3D"http://www.api-digital.c=
om/" target=3D"_blank">http://www.api-digital.com</a> --<br>New to Asterisk=
? Join us for a live introductory webinar every Thurs:<br>=A0 =A0 =A0 =A0 =
=A0 =A0 =A0 <a href=3D"http://www.asterisk.org/hello" target=3D"_blank">htt=
p://www.asterisk.org/hello</a><br>
<br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<=
br>=A0 <a href=3D"http://lists.digium.com/mailman/listinfo/asterisk-users" =
target=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</=
a><br>
</div></div></blockquote></div><br>
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