Hi Aqs,
rtpproxy should start relaying immediately to the port specified in the
SDP. If the UA sends UDP packets from another source:IP port, it should
update internal session data, so that all subsequent packets will be
relayed to the proper IP:port. Otherwise it will continue to user the
same IP:port specified in the SDP.
Is the IP in the SDP negotiation correct (the public one, and not the
local one) ?
Have you had a look at the rtpproxy log ?
Daniel
On 05/18/2018 10:37 AM, Abdul Basit wrote:
It looks a simple question but whats outgoing RTP packets destination?
Is CALLEE IP present on that NATed destination?
Confirm that RTP Proxy determined that NATed IPs properly.
What did you receive in 200OK from CALLEE?
Check if engaged RTP ports are open in the flow?
--
regards,
abdul basit
On 18 May 2018 at 13:08, Bilal Abbasi <bilaln...@gmail.com
<mailto:bilaln...@gmail.com>> wrote:
Corrigendum:
We are getting packets from CALLER*
Regards
Abbasi
On Fri, 18 May 2018 at 12:46 PM, Bilal Abbasi <bilaln...@gmail.com
<mailto:bilaln...@gmail.com>> wrote:
Actually they dont even get out of rtpproxy, we cant see that
in the in sngrep/tcpdump.
We are getting packets from callee but nothing going out of
rtpproxy(talking about local dump)
Regards
Abbasi
On Fri, May 18, 2018, 12:13 Giovanni Tommasini - evosip
<giovanni.tommasini@evosip.cloud> wrote:
Hi Younas,
when you have a call if you make a trace in your sip
serverwith sngrep or tshark can you see the RTP packet ?
you says "I have kamailio server behind nat with
rptproxy.", so is it possible that the router in front
your SIP server blocks the traffic? could you have a trace
there?
I mean the packets exit the rtpproxy but don't arrive to
callee or just don't come out of the rtpproxy?
Giovanni Tommasini | *
evosip* <http://evosip.cloud>
2018-05-18 1:36 GMT+02:00 Aqs Younas <aqsyou...@gmail.com
<mailto:aqsyou...@gmail.com>>:
Thanks for replying. Callee is actually a sip
provider. I see no firewall or iptables rules on
server to prevent rtp following towards callee.
Is there anything i can do to see which thing is
blocking rtps.
Best
On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks
<a...@goflyball.com <mailto:a...@goflyball.com>> wrote:
Is there something blocking RTP traffic from
reaching the callee? Is the callee a carrier or
the actually endpoint endpoint (aka SIP Phone)?
*Mack Hendricks / Head of Support / dOpenSource*
web: http://dopensource.com
support: +888-907-2085
dSIPRouter <http://dsiprouter.org> - GUI focused
on implementing Kamailio to provide SIP Trunking
and PBX Hosting Services
On May 18, 2018, at 12:50 AM, Aqs Younas
<aqsyou...@gmail.com
<mailto:aqsyou...@gmail.com>> wrote:
Greetings list,
I have kamailio server behind nat with rptproxy.
But i am getting no voice on the call. After
taking trace i could see that rtpproxy was
getting rtp packets but not packets was being
forwarded towards callee side.
Though i see in rtpproxy logs, packets being
relayed from caller side.
Does rtpproxy wait to receive a single rtp frame
from callee before fowarding that to callee?
Since, i am getting no rtp from callee. So, might
be rptrpoxy is waiting to learn source address of
callee side rtp, that is why it is not forwarding
rtp packets from caller towards callee.
Any pointer/suggestion is much appreciated.
Best Regards,
Aqs Younas
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