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 <bilaln018@gmail.com> wrote:
Corrigendum:
We are getting packets from CALLER*

Regards
Abbasi

On Fri, 18 May 2018 at 12:46 PM, Bilal Abbasi <bilaln018@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 server with 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

2018-05-18 1:36 GMT+02:00 Aqs Younas <aqsyounas@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 <ap@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
support: +888-907-2085
dSIPRouter - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

On May 18, 2018, at 12:50 AM, Aqs Younas <aqsyounas@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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