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https://github.com/onmyway133/rtpproxy

🥂 Forked from http://www.rtpproxy.org/ with IP handover support
https://github.com/onmyway133/rtpproxy

handover proxy rtp

Last synced: 4 months ago
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🥂 Forked from http://www.rtpproxy.org/ with IP handover support

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README 

        
I. About



The RTPproxy is a high-performance software proxy for RTP streams that can

work together with SER or OpenSER. 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 now is being actively maintained by the Sippy Software, Inc.



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 SER mailing lists

.



Commercial support is available from the Sippy Software, Inc. - visit

http://www.sippysoft.com for details.



$Id: README,v 1.3 2007/07/28 01:55:12 sobomax Exp $