/rtpengine

The Sipwise media proxy for Kamailio

Primary LanguageCGNU General Public License v3.0GPL-3.0

Code Testing Debian Package CI Coverity

What is rtpengine?

The Sipwise NGCP rtpengine is a proxy for RTP traffic and other UDP based media traffic. It's meant to be used with the Kamailio SIP proxy and forms a drop-in replacement for any of the other available RTP and media proxies.

Currently the only supported platform is GNU/Linux.

Mailing List

For general questions, discussion, requests for support, and community chat, join our mailing list. Please do not use the Github issue tracker for this purpose.

Features

  • Media traffic running over either IPv4 or IPv6
  • Bridging between IPv4 and IPv6 user agents
  • Bridging between different IP networks or interfaces
  • TOS/QoS field setting
  • Customizable port range
  • Multi-threaded
  • Advertising different addresses for operation behind NAT
  • In-kernel packet forwarding for low-latency and low-CPU performance
  • Automatic fallback to normal userspace operation if kernel module is unavailable
  • Support for Kamailio's rtpproxy module
  • Legacy support for old OpenSER mediaproxy module
  • HTTP, HTTPS, and WebSocket (WS and WSS) interfaces

When used through the rtpengine module (or its older counterpart called rtpproxy-ng), the following additional features are available:

  • Full SDP parsing and rewriting
  • Supports non-standard RTCP ports (RFC 3605)
  • ICE (RFC 5245) support:
    • Bridging between ICE-enabled and ICE-unaware user agents
    • Optionally acting only as additional ICE relay/candidate
    • Optionally forcing relay of media streams by removing other ICE candidates
    • Optionally act as an "ICE lite" peer only
  • SRTP (RFC 3711) support:
    • Support for SDES (RFC 4568) and DTLS-SRTP (RFC 5764)
    • AES-CM and AES-F8 ciphers, both in userspace and in kernel
    • HMAC-SHA1 packet authentication
    • Bridging between RTP and SRTP user agents
    • Opportunistic SRTP (RFC 8643)
    • Legacy non-RFC (dual m= line) best-effort SRTP
    • AES-GCM Authenticated Encryption (AEAD) (RFC 7714)
    • a=tls-id as per RFC 8842
  • Support for RTCP profile with feedback extensions (RTP/AVPF, RFC 4585 and 5124)
  • Arbitrary bridging between any of the supported RTP profiles (RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF)
  • RTP/RTCP multiplexing (RFC 5761) and demultiplexing
  • Breaking of BUNDLE'd media streams (draft-ietf-mmusic-sdp-bundle-negotiation)
  • Recording of media streams, decrypted if possible
  • Transcoding and repacketization
  • Transcoding between RFC 2833/4733 DTMF event packets and in-band DTMF tones (and vice versa)
  • Injection of DTMF events or PCM DTMF tones into running audio streams
  • Playback of pre-recorded streams/announcements
  • Transcoding between T.38 and PCM (G.711 or other audio codecs)
  • Silence detection and comfort noise (RFC 3389) payloads
  • Media forking
  • Publish/subscribe mechanism for N-to-N media forwarding

There is also limited support for rtpengine to be used as a drop-in replacement for Janus using the native Janus control protocol (see below).

Rtpengine does not (yet) support:

  • ZRTP, although ZRTP passes through rtpengine just fine

Documentation

Check our general documentation here:

For quick access, documentation for usage:

For quick access, documentation for development:

Sponsors

Contribution

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