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 <[email protected]
<mailto:[email protected]>> 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
<[email protected] <mailto:[email protected]>> 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
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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
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M. Salman Zafar
VoIP Professional
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