[Users] two invites, different session description => broken

[Olle E Johansson]

30 nov 2006 kl. 10.42 skrev Bogdan-Andrei Iancu:

> Hi John,
>
> actually if you use Asterisk, there is no need for using RTPproxy  
> as Asterisk is able to cope with nated rtp by itslef (using Comedia).
>
Depends on the setup really, Bogdan.

If your devices are registering with OpenSER, you need RTP proxy. If  
they're registering with Asterisk and calling through
Asterisk, asterisk can handle media and NAT.

If you have NATs, you should really disable can-reinvites since you  
don't want ASterisk to set up media stream
that will fail.

/O

> regards,
> bogdan
>
> John Peters wrote:
>
>> ONsip has some tips for handling re-INVITEs with rtpproxy:
>>
>> http://siprouter.onsip.org/doc/gettingstarted/ 
>> ch08s02.html#rtp_loose_route <http://siprouter.onsip.org/doc/ 
>> gettingstarted/ch08s02.html#rtp_loose_route>
>>
>> Advises to use force_rtp_proxy(l) on reinvites.
>>
>> On 11/29/06, *John Peters* <petersprc at gmail.com  
>> <mailto:petersprc at gmail.com>> wrote:
>>
>>     Not sure why that's happening. Probably setting canreinvite=no on
>>     the asterisk side will eliminate the re-INVITEs as a temporary
>>     solution, but still would like to know what is happening...
>>
>>     wrote:
>>     > Sometimes, a calls b and b hears a, and a hears b for a second
>>     but a second
>>     > INVITE comes to phone B that causes it to redirect rtp to be
>>     point to point.
>>     > Sometimes there is no audio.
>>     > Sometimes, everything works fine.
>>
>>     > At one point, rtp from a was going to asterisk, but asterisk  
>> was
>>     not sending
>>     > the rtp on to b, and b was trying to send traffic point to  
>> point.
>>
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>
>
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* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden