[OpenSIPS-Users] OpenSIPS/RTpProxy BridgeMode after failure route

[Răzvan Crainea]

Hi, Ali!

For the initial branch (in request route) are you using 
engage_rtpproxy()? If so, try to use rtpproxy_offer().

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 10/09/2014 12:06 AM, Ali Pey wrote:
> Hello Salman,
>
> Can you please elaborate on how you got this working? I have the same 
> problem and can't get it to work.
>
> In failure route, I do:
> unforce_rtp_proxy()
> Then when I have a new destination, I do:
> rtpproxy_offer("rocie");
>
> However, I end up with messed up SDP, in my second invite. It doesn't 
> remove the old IP addresses and only adds the IP addresses again:
> o=Sonus_UAC 9216 20203 IN IP4 10.160.11.16210.160.11.162a Capabilities
> c=IN IP4 10.160.11.16210.160.11.162udio 2311822970AVP 0 8 100
>
>
> Please let me know how I can fix this.
>
> Thanks.
>
>
> On Mon, Jan 6, 2014 at 10:26 AM, Salman Zafar <msalman212 at gmail.com 
> <mailto:msalman212 at gmail.com>> wrote:
>
>     Hi Razvan,
>             I got it working without branching, after banging head a
>     lot I got to learn unforcing drops the media ports for previous
>     rtpproxy offer/answer and after that directing the new flow though
>     rtpproxy flags,IP media works. I am able to traverse from eternal
>     to internal play media and then on failure do external to external
>     with media flowing between public interfaces. Just wondering if
>     you know this method or certify.
>
>
>
>     On Mon, Jan 6, 2014 at 4:35 PM, Răzvan Crainea
>     <razvan at opensips.org <mailto:razvan at opensips.org>> wrote:
>
>         Hi, Salman!
>
>         The sockets used by RTPProxy are created when the session is
>         started (the first offer) and cannot be updated afterwards.
>         Therefore the only solution I can see is to configure a per
>         branch scenario, as you mentioned.
>
>         Best regards,
>
>         Razvan Crainea
>         OpenSIPS Core Developer
>         http://www.opensips-solutions.com
>         <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=c1574c01-908b-4910-aaff-83f9f8f63efd&cm_destination=http://www.opensips-solutions.com>
>
>
>
>         On 12/30/2013 01:11 PM, Salman Zafar wrote:
>
>             Hi,
>                 I have a scenario of playing media at a private-ip
>             media server and
>             send BUSY, next in failure route bridge call to a public
>             IP. (SIP to SIP).
>
>             So the scenario is as follows:
>
>             UA(Phone1) -> OpenSIPS/RTpProxy(ei) -> Media-Server
>             (Private IP) -> BUSY
>             -> OpenSIPS(failure route) -> RTpProxy(ee) -> lookup ->
>             (UA Phone2)
>
>             Now the problem is RtpProxy is being offered (EI flags) in
>             first case
>             where routing to Media sever at private IP, after failure
>             it is again
>             used with (EE flags), also in corresponding replies.
>
>             The second time RTpProxy does not effect SDP c= and ports
>             in a way to
>             build media communication. SDP fix directly does not
>             effect rtp ports.
>
>             Is there any way of using RtpProxy differently in
>             fail-over, or I have
>             to go for rtpproxy per branch?.
>
>
>             Thanks in advance.
>
>             --
>             Regards
>
>             Salman
>
>
>
>             _______________________________________________
>             Users mailing list
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>             http://lists.opensips.org/cgi-
>             <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=0138242a-38a8-4160-8e04-9e778d8a3ff2&cm_destination=http://lists.opensips.org/cgi->bin/mailman/listinfo/users
>
>
>         _______________________________________________
>         Users mailing list
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>         http://lists.opensips.org/cgi-
>         <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=39ab01b6-8245-48af-9ab5-ed7494a6765b&cm_destination=http://lists.opensips.org/cgi->bin/mailman/listinfo/users
>
>
>
>
>     -- 
>     Regards
>
>     M. Salman Zafar
>
>     VoIP Professional
>
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>     <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=8efee758-5809-4052-89d5-9190de57c58c&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20141009/a755ce46/attachment-0001.htm>