asterisk users - Jun 2014 - Renegotiate SIP audio codec after call is up

<div style="font:14px/1.5 'Lucida Grande',
'微软雅黑';color:#333;"><p style="line-height: 1.5; margin:
0px; font-family: 'Lucida Grande', 'Lucida Sans Unicode',
sans-serif !important;">Hi All,</p><p style="line-height:
1.5; margin: 0px; font-family: 'Lucida Grande', 'Lucida Sans
Unicode', sans-serif !important;">Asterisk from 11.X branch is able
to renegotiate an audio codec after a <span style="color: rgb(0, 0, 0);
line-height: normal;">SIP call session has been established (INVITE and
200 OK)?</span></p><p style="line-height: 1.5; margin: 0px;
font-family: 'Lucida Grande', 'Lucida Sans Unicode', sans-serif
!important;"><span style="color: rgb(0, 0, 0); line-height:
normal;"><br></span></p><p style="margin: 0px;
font-family: 'Lucida Grande', 'Lucida Sans Unicode', sans-serif
!important;"><span style="line-height: normal; color: rgb(0, 0,
0);">I have a problem with a reinvite sent by our SIP provider to change
audio </span><span style="line-height: normal; color: rgb(0, 0,
0);">codec due to the recognition of a fax tone: after the call is
</span><span style="color: rgb(0, 0, 0); line-height:
normal;">established in g729, after a while </span><span
style="color: rgb(0, 0, 0); line-height: normal;">I have the
reinvite sent by the SIP provider with g711 in </span><span
style="color: rgb(0, 0, 0); line-height: normal;">the SDP; Asterisk
(v 1.4.33.1) says "</span><span style="color: rgb(0, 0, 0);
line-height: normal;">Oooh, we need to change our audio formats since
our peer supports only </span><span style="color: rgb(0, 0, 0);
line-height: normal;">g729" and sends back 200 OK to the provider;
at this point I have one no audio.</span></p><p
style="margin: 0px; font-family: 'Lucida Grande', 'Lucida Sans
Unicode', sans-serif !important;"><span style="color: rgb(0,
0, 0); line-height: normal;"> </span></p><p
style="margin: 0px; font-family: 'Lucida Grande', 'Lucida Sans
Unicode', sans-serif !important;"><span style="color: rgb(0,
0, 0); line-height: normal;">So it seems that Asterisk responds 200 OK
to the reinvite but really can not change the codec.</span></p><p
style="margin: 0px; font-family: 'Lucida Grande', 'Lucida Sans
Unicode', sans-serif !important;"><font
color="#000000"><span style="line-height:
normal;">Is that correct?</span></font></p><p
style="line-height: 1.5; margin: 0px; font-family: 'Lucida Grande',
'Lucida Sans Unicode', sans-serif
!important;"><br></p><p style="line-height: 1.5;
margin: 0px; font-family: 'Lucida Grande', 'Lucida Sans
Unicode', sans-serif !important;">Best regards,</p><p
style="line-height: 1.5; margin: 0px; font-family: 'Lucida Grande',
'Lucida Sans Unicode', sans-serif
!important;">Matteo</p></div>

How many g729 Licenses do you have?

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Matteo Campana
Sent: Wednesday, June 04, 2014 10:48 AM
To: asterisk-users
Subject: [asterisk-users] Renegotiate SIP audio codec after call is up


Hi All,

Asterisk from 11.X branch is able to renegotiate an audio codec after a SIP call
session has been established (INVITE and 200 OK)?



I have a problem with a reinvite sent by our SIP provider to change audio codec
due to the recognition of a fax tone: after the call is established in g729,
after a while I have the reinvite sent by the SIP provider with g711 in the SDP;
Asterisk (v 1.4.33.1) says "Oooh, we need to change our audio formats since
our peer supports only g729" and sends back 200 OK to the provider; at this
point I have one no audio.



So it seems that Asterisk responds 200 OK to the reinvite but really can not
change the codec.

Is that correct?



Best regards,

Matteo
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