No subject
Thu Jul 10 23:27:17 CEST 2014
failover/failure_route your call goes to second Carrier and since the
use_media_proxy has been called for first one, hand shake has been done in
SDP the media relay still waits on for the first Media stream to continue.
What you really want is that even in case of rollover to second , third, or
even fourth carrier the media-proxy should redo the SDP exchange for each
carrier failover.
For the above mentioned scenario you need to end_media_proxy() (possibly in
failure_route), then in the config area where you dial over to second
gateway recall the use_media_proxy function so that media proxy be ready to
exchange media with newer SDP IPs.
BTW if the far end has "rejected" the call by sending Busy you shouldn't
loop try dial again, its not good to keep ringing someone once they've
rejected call; only if the carrier has sent you some other SIP failure code
only then you should go for re-attempt.
Regards,
On Fri, Nov 7, 2014 at 7:09 AM, Kristian F. H=C3=B8gh <kfh.opensips at kfh.dk>
wrote:
> Hi,
>
> I receive an INVITE from a NAT'ed client, which I forward to a second
> proxy after calling use_media_proxy.
> The second proxy send the INVITE upstream.
> Upstream starts sending RTP (to media-relay), then rejects the call.
> I try next gateway (on second proxy), which answers 100, 183 and later 20=
0
> OK. (And sends RTP to media-relay)
>
> Unfortunately the relay received 7 RTP packages from the failed attempt,
> and therefore RTP from succesfull call is ignored.
>
> As I know the IP range is non- NAT'ed, I would like to ignore the RTP fro=
m
> the first gateway, by forcing relay to use SDP.
> Is it posible to tell the relay "This client is not NAT'ed, use IP
> address/port from SDP"?
> Or
> Don't forward RTP before SDP. (That way, RTP from failed call would be
> ignored)
>
> Regards,
> Kristian H=C3=B8gh
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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<div dir=3D"ltr">Hi,<div><br><div>Here is what it looks like to me;=C2=A0</=
div><div><div>NAT'd Client<=3D=3D=3D=3D=3D>OpenSIPS+Media-Proxy&l=
t;=3D=3D=3D=3D=3D=3D>Second Proxy??<=3D=3D=3D=3D=3D>Upstream</div>=
<div><br></div><div>From what I understand, your main issue is that after f=
ailover/failure_route your call goes to second Carrier and since the use_me=
dia_proxy has been called for first one, hand shake has been done in SDP th=
e media relay still waits on for the first Media stream to continue. =C2=A0=
</div><div><br></div><div>What you really want is that even in case of roll=
over to second , third, or even fourth carrier the media-proxy should redo =
the SDP exchange for each carrier failover.</div><div><br></div><div>For th=
e above mentioned scenario you need to end_media_proxy() (possibly in failu=
re_route), then in the config area where you dial over to second gateway re=
call the use_media_proxy function so that media proxy be ready to exchange =
media with newer SDP IPs.</div><div><br></div><div>BTW if the far end has &=
quot;rejected" the call by sending Busy you shouldn't loop try dia=
l again, its not good to keep ringing someone once they've rejected cal=
l; only if the carrier has sent you some other SIP failure code only then y=
ou should go for re-attempt. =C2=A0<br></div></div><div><br></div><div><br>=
</div><div>Regards,</div><div><br></div></div></div><div class=3D"gmail_ext=
ra"><br><div class=3D"gmail_quote">On Fri, Nov 7, 2014 at 7:09 AM, Kristian=
F. H=C3=B8gh <span dir=3D"ltr"><<a href=3D"mailto:kfh.opensips at kfh.dk" =
target=3D"_blank">kfh.opensips at kfh.dk</a>></span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">Hi,<br>
<br>
I receive an INVITE from a NAT'ed client, which I forward to a second p=
roxy after calling use_media_proxy.<br>
The second proxy send the INVITE upstream.<br>
Upstream starts sending RTP (to media-relay), then rejects the call.<br>
I try next gateway (on second proxy), which answers 100, 183 and later 200 =
OK. (And sends RTP to media-relay)<br>
<br>
Unfortunately the relay received 7 RTP packages from the failed attempt, an=
d therefore RTP from succesfull call is ignored.<br>
<br>
As I know the IP range is non- NAT'ed, I would like to ignore the RTP f=
rom the first gateway, by forcing relay to use SDP.<br>
Is it posible to tell the relay "This client is not NAT'ed, use IP=
address/port from SDP"?<br>
Or<br>
Don't forward RTP before SDP. (That way, RTP from failed call would be =
ignored)<br>
<br>
Regards,<br>
Kristian H=C3=B8gh<br>
<br>
<br>
_______________________________________________<br>
Users mailing list<br>
<a href=3D"mailto:Users at lists.opensips.org">Users at lists.opensips.org</a><br=
>
<a href=3D"http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target=
=3D"_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br=
>
</blockquote></div><br></div>
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