Showing posts with label rtpproxy. Show all posts
Showing posts with label rtpproxy. Show all posts

Friday, August 23, 2013

New Module: rtpproxy-ng – WebRTC to RTP

The GIT master branch of Kamailio includes now a new module – rtpproxy-ng. It is designed to be next generation RTP relay control protcol, using bencode as the base for formatting control command. It can be used as a drop-in replacement for old rtpproxy module, but you have to use mediaproxy-ng as RTP relay. Both rtpproxy-ng module and mediaproxy-ng application were developed by Sipwise, main author in the Kamailio devel team being Richard Fuchs.
One of the very appealing features when using rtpproxy-ng and mediaproxy-ng is the ability to bridge WebRTC endpoints to classic SIP phones without any dedicated SBC or media gateway. Mediaproxy-ng is able to decode/encode SRTP to RTP back and forth.
For more details about this module, read:
Mediaproxy-ng can be found at:

Monday, September 29, 2008

Benchmark: RTPProxy v1.0 with OpenSER 1.3

Transnexus has done an interesting analysis of using RTPProxy 1.0 with OpenSER 1.3, of course specific with their use case.

It has very interesting results, you can check here:

http://www.transnexus.com/White Papers/OpenSER_RTPproxy_test.htm

There was a lot of discussing regarding space for improvements and tunings to get better performances on the mailing lists, but still it is a good reference to be used. The page includes details about everything you need to do the test by yourself.

Transnexus is the developer of OSP module in Kamailio/OpenSER.

Tuesday, June 24, 2008

Updates of NAT Traversal Solutions for OPENSER

There were lot of development efforts in the two options available for NAT traversal with OPENSER.

Mediaproxy relesed next generation version: v2.0. More details at:

http://ag-projects.com/MediaProxy.html

Rtpproxy brings lately lot of new stuff, information is available at the dedicated site:

http://www.rtpproxy.org