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RTPProxy - application for RTP packets relaying - additional patches

License: BSD 2-Clause "Simplified" License

Shell 38.52% C++ 28.49% C 32.99%

rtpproxy's Introduction

I. About

The RTPproxy is a high-performance software proxy for RTP streams that can
work together with SER, OpenSER or Kamailio. Originally created for handling NAT
scenarious it can also act as a generic media relay as well as gateway RTP
sessions between IPv4 and IPv6 networks. RTPproxy was developed by Maxim
Sobolev and maintained by the Sippy Software, Inc.

This is a clone that keeps the original source code and adds several patches
in top of it.

The RTPproxy supports some advanced features, such as remote control mode,
allowing building scalable distributed SIP VoIP networks. The nathelper module
included into the SER or OpenSER SIP Proxy software allows using multiple
RTPproxy instances running on remote machines for fault-tolerance and
load-balancing purposes.

The software also supports video relaying and RTP session recording.


II. How it works

This proxy works as follows:

- When SER receives INVITE reqiest, it extracts call-id from it and
  communicates it to the proxy via Unix domain socket. 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
  empty UDP port from the range specified at the compile time and returns
  number of that port to a SER. After receiving reply from the proxy, SER
  replaces media ip:port in the SDP to point to the proxy and forwards
  reqiest as usually;

- when SER receives non-negative SIP reply with SDP it again extracts
  call-id 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, SER 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).


III. Limitations

Currently, rtpproxy does not support command-line definition of RTP port
range. By default rtpproxy uses UDP ports 35000-65000 and these should be
opened any firewalls in front of rtpproxy. The ports used can be modified in
rtpp_defines.h file (PORT_MIN and PORT_MAX) and then recompile/reinstall.


IV. TODO (in no particular order)

- Port to another OSes;

- make more parameters (e.g. ports range, max idle time etc.) to be
  adjustible via command line.


V. Support

Community-based support could be obtained via Kamailio mailing lists
<[email protected]>.

VI. Initial Development

This software started as clone from RTPProxy developed by Sippy Software, Inc.,
see:
- http://www.rtpproxy.org

rtpproxy's People

Contributors

sobomax avatar lemenkov avatar miconda avatar

Watchers

James Cloos avatar

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