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

[Bogdan-Andrei Iancu]

Hi Olle,

even if the devices register with OpenSER and you use Asterisk as B2BUA 
behind the proxy, it still works if:
    1) handle SIP signalling NAT mangle in OpenSER
    2) disable the can-reinvites option in Asterisk.
    3) do not use rtpproxy.

 From Asterisk perspective, do you see any issues in this scenario?

thanks and regards,
bogdan

Olle E Johansson wrote:

>
> 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
>
>
>