[SR-Users] rtpproxy module question

Daniel-Constantin Mierla miconda at gmail.com
Fri May 27 11:01:56 CEST 2016


Hello,

I am not familiar with the insights of rtpproxy source code, so I don't
know if there is a limitation for duration and for how long it is.

Cheers,
Daniel


On 27/05/16 10:42, gmele wrote:
>
> 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]
> </user/SendEmail.jtp?type=node&node=148854&i=0>]
> *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
>
>  
>
> On 27/05/16 09:55, gmele wrote:
>
>     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
>
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>
>
> -- 
> Daniel-Constantin Mierla
> http://www.asipto.com - http://www.kamailio.org
> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda
>
>
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-- 
Daniel-Constantin Mierla
http://www.asipto.com - http://www.kamailio.org
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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