Hello,

 

Thx for the explanation, so it means that as soon as the callee connects to the RTP Proxy, the rtp proxy will use the callee ip address and port to forward the rtp stream and ignore the initial learned ip/port? Is there a duration limitation in this learning mode? Meaning that if the callee waits to much to send the first udp packet, the rtp proxy will use the ip/port set during negotiation?

 

Thx

 

Giovanni

 

From: Daniel-Constantin Mierla-6 [via SIP Router] [mailto:ml-node+[hidden email]]
Sent: vendredi 27 mai 2016 10:09
To: Mele Giovanni
Subject: Re: rtpproxy module question

 

Hello,

initially the rtpproxy is in so called learning mode, waiting for the first rtp packet to come from each side of the call. Before receiving first rtp packet it relies on source ip of signaling.

If the SDP has the device IP (you can eventually set that in the proxy), then you can use 'r' flag for rtpproxy_manage() to tell rtpproxy that it should trust the IP from sdp.

Cheers,
Daniel

 

Hello,

we are using an RTP Proxy from rtpproxy.org as media relay to establish

communication between our mobile phones. Of course, we are using the

kamailio rtpproxy module to modify the SDP payload and control the proxy.

In our Kamailio configuration, we have 1 kamailio configured as Proxy and

one kamailio configured as Registrar. So calls go through the Proxy and then

to the Registrar who will update the SDP header and select an available rtp

proxy.

We have noticed that sometimes, the rtp udp flow between the phones isn't

routed properly by the rtpproxies, ending in the communication drop (all the

SIP nego is working well, and the SDP are correctly patched with the rtp

proxy address and port). 

Analyzing the RTP proxy packets, we have found that the Kamailio registrar

gives the Kamailio proxy ip address in the RTP proxy create session command,

but keeps the original sdp port.

command looks like this:

Uc96,101 DC -PbO~Wnm <proxy_ip> <port from original sdp phone packet>

PZU5OITCW;1 

We are using rtpproxy_manage() without any flags.

It seems to us that this ip and port are used as default forward route as

long as the callee hasnt connected to the rtpproxy. *Is it correct?

*

If its true, Its seems to us that this cant work as we are mixing the proxy

sip address with a udp port open on the phone ? *Is our analysis correct ?*

Can we use some option in rtpproxy_manage to replace the proxy ip by the

phone ip as seen in the via route ? 

Thx for your help

Giovanni

--

View this message in context: http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.html

Sent from the Users mailing list archive at Nabble.com.

_______________________________________________

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list

[hidden email]

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 

Daniel-Constantin Mierla

http://www.asipto.com - http://www.kamailio.org

http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
[hidden email]
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users