Hello All! 2013/1/2 Klaus Darilion <[email protected]>:
> I just heard of erlrtpproxy for the first time. There is pretty no > documentation in github. Is it comatible with rtpproxy. Are there any > advantages over rtpproxy? > > I would be cool of you can shortly present erlrtpproxy to the Kamailio > users. I recently picked up my old attempt (which was in a quite good state already) to rewrite a venerable RTPproxy in Erlang. Right now it still in a w.i.p. although I'm using it personally in a two installations. * https://github.com/lemenkov/erlrtpproxy What I've done so far is: * Basic offer/answer works. * Call recording works * Music-on-hold works. * Transcoding from/to G.711 / G.721 / G.722 / G.723 / G.726 / G.729 (annex A only, annex B VAD frames currently skipped) / GSM / iLBC / Opus / SPEEX. I made the first version to be compatible with a very old Erlang so it works sub-optimally on a more recent Erlang versions (speaking more positively this means I have a room for improvements). * OpenSER-style timeout notifications via IPv4/IPv6 UDP/TCP (configurable) * RADIUS notifications (start / interim update / stop, configurable) * Control socket (22222 by default) using IPv4/IPv6 UDP/TCP (configurable) * Events logging (start, stop, interim update, timeout, codec/parser error) to syslog (with full RTCP pretty printing) * WebRTC-style RTP/RTCP muxing/demuxing from/to two independent channels (automatically) * Setting some internal parameters via HTTP-JSON interface. * IPv4/IPv6 bridging for RTP/RTCP * JSON stats retrieval via HTTP (currently supporting queries like http://localhost:8442/json/[email protected]&payload=0 - available parameters are - payload, callid, mediaid, tag, remote IP) {"http_query":{"callid":"[email protected]","payload":"0"},"result":{"media":{"callid":"[email protected]","mediaid":"1","tag":"0003e30cc50c5d43416d30b4-7b2f969c","payload":0,"rxbytes":108008800,"rxpackets":677772,"txbytes":108262240,"txpackets":676639,"local":{"ip":"192.168.1.233","rtp":50486,"rtcp":50487},"remote":{"ip":"192.168.5.1","rtp":5038,"rtcp":5039}},"media":{"callid":"[email protected]","mediaid":"1","tag":"2e268de3","payload":0,"rxbytes":108262240,"rxpackets":676639,"txbytes":108008800,"txpackets":677772,"local":{"ip":"192.168.1.233","rtp":34732,"rtcp":34733},"remote":{"ip":"192.168.5.10","rtp":20912,"rtcp":null}}}} In progress * A completely new JSON-based command protocol. * Multiple networks bridging (more than just three - internal, external, ipv6). * STUN/TURN/ICE. * More stats and more query options (for example, select all calls from/to a given country, using GeoIP) * SRTP / ZRTP (almost there - I put them on hold for now) and a gateway SRTP<->RTP * Video with transcoding. * Full WebRTC media support. * CEE logging. * Stream copying to a given IP:Port (actually I'm not sure that I'm going to implement this feature). I won't plan to add: * Transcoding offloading using dedicated hardware. I just don't have any hardware available. * In-kernel routing (like iptrtpproxy, mediaproxy did). Since I'm using Erlang I've got for free a lot of good stuff - clustering, fault tolerance, restarts in case of failures, almost linear scalability, etc. It has few disadvantages as well. Most notable one - Erlang isn't a very friendly thing especially when it comes to the installation. I'm working on improving this. Regarding performance - ~1000 concurrent calls with G.711 codec and without transcoding costs ~500% utilization of a 8-core 32-bit x86 2GHz CPU which isn't a very good but I'm working on the improvement. -- With best regards, Peter Lemenkov. _______________________________________________ sr-dev mailing list [email protected] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
