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https://github.com/sippy/rtpproxy

The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.
https://github.com/sippy/rtpproxy

kamailio opensips rtp sip sippy

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The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.

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## About



The RTPproxy is a extremely reliable and reasonably high-performance software

proxy for RTP streams that can work together with [OpenSIPS](https://opensips.org),

[Kamailio](https://kamailio.org) or [Sippy B2BUA](https://github.com/sippy/b2bua).



Originally created for handling NAT scenarios, back in 2004-2005, it can also act

as a generic real time datagram relay as well as gateway Real-Time Protocol (RTP)

sessions between IPv4 and IPv6 networks.



The RTPproxy supports many advanced features and is controllable over

multitude of Layer 4 protocols, including Unix Domain, UDP, UDPv6, TCP and TCPv6.



The software allows building scalable distributed SIP networks. The rtpproxy module

included into the OpenSIPS or Kamailio SIP Proxy software allows using multiple

RTPproxy instances running on remote machines for fault-tolerance and

load-balancing purposes.



Advanced high-capacity clustering and load balancing is available through the

use of [RTP Cluster](https://github.com/sippy/rtp_cluster) middleware.



The software also supports MOH/pre-recorded media injection,  video relaying

and session recording to a local file or remote UDP listener(s). As well

as makes available array of real-time or near real-time session counters,

both per-session and per-instance.



Since version 3.1.0, full set of extensions is available allowing to create

a WebRTC-compatible endpoints.



## News



- introducing WebRTC/WSS clients support via 3 new modules: dtls_gw, ice_lite

  and rtcp_demux.



## How it works



This proxy works as follows:



- When SIP Controller, either proxy or B2BUA, receives INVITE request, it

  extracts call-id from it and communicates it to the proxy via control

  channel. Proxy looks for an existing sessions with such id, if the session

  exists it returns UDP port for that session, if not, then it creates a new

  session, binds to a first available randomly selected pair of UDP ports and

  returns number of the first port. After receiving reply from the proxy, SIP

  Controller replaces media ip:port in the SDP to point to the proxy and

  forwards request as usually;



- when SIP Controller receives non-negative SIP reply with SDP it again

  extracts call-id along with session tags from it and communicates it to

  the proxy. In this case the proxy does not allocate a new session if it

  doesn't exist, but simply performs a lookup among existing sessions and

  returns either a port number if the session is found, or error code

  indicating that there is no session with such id. After receiving positive

  reply from the proxy, SIP Controller replaces media ip:port in the SIP

  reply to point to the proxy and forwards reply as usually;



- after the session has been created, the proxy listens on the port it has

  allocated for that session and waits for receiving at least one UDP

  packet from each of two parties participating in the call. Once such

  packet is received, the proxy fills one of two ip:port structures

  associated with each call with source ip:port of that packet. When both

  structures are filled in, the proxy starts relaying UDP packets between

  parties;



- the proxy tracks idle time for each of existing sessions (i.e. the time

  within which there were no packets relayed), and automatically cleans

  up a sessions whose idle times exceed the value specified at compile

  time (60 seconds by default).



## Building from github



```

$ git clone -b master https://github.com/sippy/rtpproxy.git

$ git -C rtpproxy submodule update --init --recursive

$ cd rtpproxy

$ ./configure

$ make

```



## Authors and Contributors



The RTPproxy has been designed by Maxim Sobolev and now is being actively

maintained by the [Sippy Software, Inc](http://www.sippysoft.com). With the

great help of numerous community contributors, both private and institutional.

Not to mention army of robots gracefully dispatched at need by CI.



The original idea has inspired and directly influenced multitude of independent

implementations, including but not limited to the

[Mediaproxy](https://mediaproxy.com),

[erlrtpproxy](https://github.com/lemenkov/erlrtpproxy), and most recently

[RTP Engine](https://github.com/sipwise/rtpengine), each project focusing on

its own area of the vast functionality space.



## Documentation and References



- [User Manual](https://www.rtpproxy.org/doc/master/user_manual.html#MAKESRC);

- [RTPproxy Control Protocol Reference](https://github.com/sippy/rtpproxy/wiki/RTPProxy-Command-Protocol).



## Feedback & Support



Open a ticket on the github issue tracker, or post a message on the [mailing

list](https://groups.google.com/forum/#!forum/rtpproxy)