The RTPproxy is a extremely reliable and reasonably high-performance software proxy for RTP streams that can work together with OpenSIPS, Kamailio or 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 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.
- introducing WebRTC/WSS clients support via 3 new modules: dtls_gw, ice_lite and rtcp_demux.
This proxy works as follows:
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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;
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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;
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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;
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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).
$ git clone -b master https://github.com/sippy/rtpproxy.git
$ git -C rtpproxy submodule update --init --recursive
$ cd rtpproxy
$ ./configure
$ make
The RTPproxy has been designed by Maxim Sobolev and now is being actively maintained by the Sippy Software, Inc. 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, erlrtpproxy, and most recently RTP Engine, each project focusing on its own area of the vast functionality space.
Open a ticket on the github issue tracker, or post a message on the mailing list
