[OpenSIPS-Users] conditional call forwarding and mediaproxy

[dragos at sctp.ro]

Hi

> Hi,
>
>>
>> Yes. I am. I am sure the function gets called, because I use append_hf()
>> to add a special P-hint header in onreply_route, and I cand see it with
>> ngrep in the relay-ed reply.
>> The 'normal' calls have media through mediaproxy both ways - they are
>> okay.
>>
>
> Ok
> .
>>
>> So if I use it once it will know how to handle IP:port pair for the
>> second
>> INVITE ? (with other RURI - since I have this issue only on conditional
>> call forwarding).
>>
Yes, just the RURI. I am not sure if it has anything to do with it, but I
have 2 B2BUAs in the callflow (opensips as well). The general callflow for
a call would be:

PROVIDER->B2BUA.in(opensips)->SIP_PROXY(opensips)->
B2BUA.out(opensips)->PROVIDER.

All the mediaproxy related stuff is handled in the SIP_PROXY.

>
> If the INVITE didn't leave the proxy you shouldn't need to do anything,
> changes are made to the SDP and when you do some sort of call forwarding
> you just change the RURI, right?
>
>> Not yet. Will do eventually. Any special things to take care of for call
>> forwarding while using engage_media_proxy() ?
>>
>
> Nope, just call the function for the initial INVITE and you are all set
> :-)

If I use engage_media_proxy() it does not even replace the IP and port in
SDP for replies, though it changes them for requests.

I noticed that media-relay is closing the RTP ports when "183 Session
Progress" is received. Maybe because of the "480 - TemporarilyUnavailable"
received before ?
It happends only when using engage_media_proxy(), with use_media_proxy()
it keeps them open.

Thanks,
Dragos

>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
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