[OpenSIPS-Users] Protocol header changing with OpenSIPS

2015-06-28 Thread H Yavari
Hi,
It is possible to change the protocol header of RTP before sending and change 
it after receiving? It is possible to compress RTP packets with OpenSIPS?Any 
one can give me some point about this solutions?
Regards,H.Yavari

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Re: [OpenSIPS-Users] Protocol header changing with OpenSIPS

2015-06-28 Thread Peter Lemenkov
Short answer - no.
Long answer - it's possible with few additional components and some
additional software development. You need additional media proxy
(RTPproxy, rtpengine, Asterisk, SEMS), your own B2BUA, and some spare
time.

2015-06-28 13:32 GMT+03:00 H Yavari :
> Hi,
>
> It is possible to change the protocol header of RTP before sending and
> change it after receiving? It is possible to compress RTP packets with
> OpenSIPS?
> Any one can give me some point about this solutions?
>
> Regards,
> H.Yavari
>
>
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-- 
With best regards, Peter Lemenkov.

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Re: [OpenSIPS-Users] Transparent Auth with WebRTC

2015-06-28 Thread Satish Patel
Thanks Eric,

I tried following, now its forwarding REGISTER packet to asterisk but
authentication failed, I have check username/password is correct on
asterisk. Do you think it is because of "realm" ?

 if (is_method("REGISTER"))   {
 rewritehostport("asterisk:5060");
 route(relay);
 exit;
}




On Wed, Jun 24, 2015 at 9:24 AM, Eric Tamme  wrote:

>  just t_relay the request to your other server... OpenSIPS wont
> automatically challenge anything
>
>
> On 06/24/2015 07:22 AM, Satish Patel wrote:
>
> All,
>
>
>  I have special requirement which is little odd,  I want to use WebRTC
> with Opensips but all REGISTER process will done by other SIP server,
>
>  Example:
>
>
> [UA][WebRTC-Opensips]---[Asterisk/Freeswitch]
>
>
>  UA will use WebRTC of Opensips but opensips forward all REGISTER request
> to Asterisk/Freeswitch and user will authenticate their... In short
> Opensips will just Proxy Auth request.
>
>  How it will be possible?
>
>
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