[asterisk-users] propagate sip reinvites with directrtpsetup=yes

[Eugene Oden]

ok, thanks.  i was beginning to suspect as much but was hoping to
limit the number of components in our configuration.

thanks,

gene

On Mon, Sep 27, 2010 at 11:13 AM, Kevin P. Fleming <kpfleming at digium.com> wrote:
> On 09/27/2010 11:02 AM, Eugene Oden wrote:
>> is there a trick to get asterisk (1.6.2.13) to propagate
>> codec-changing sip reinvites when directrtpsetup=yes?
>>
>> i'm trying to route calls to a gateway without keeping asterisk in the
>> rtp stream.
>
> You are looking for a SIP proxy; Asterisk is not a SIP proxy, and no
> amount of configuration will convince it to act like one.
>
>> the gateway is first routing the call to a media server.  when
>> connecting the call to the downstream carrier a different codec is
>> selected.
>>
>> the reinvite makes it to asterisk but asterisk isn't sending it along
>> to the originator so the transmit/receive codecs are mismatched
>> causing one-way audio.
>
> Asterisk never "sends along" re-INVITEs, because the two channels
> involved in an Asterisk "call" are distinct. If the codecs are
> mismatched between the two call legs, Asterisk will try to transcode them.
>
> The 'directrtpsetup' feature is still marked *experimental*, and that is
> primarily because it defeats much of Asterisk's normal behavior; in
> addition, there a quite a few normal, working call scenarios for which
> it will fail... so it's there, but if you use it, you can expect
> difficulties.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
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