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:

• Read-the-Docs

For quick access, documentation for usage:

• Compiling and Installing
• Usage
• Transcoding
• Call recording
• The NG Control Protocol
• The TCP-NG Control Protocol
• HTTP/WebSocket support
• Janus Interface and Replacement Functionality

For quick access, documentation for development:

• Architecture Overview
• Unit-tests
• Troubleshooting Overview
• Glossary

Sponsors

• Dataport AöR

Contribution

Every bit matters. Join us. Make the rtpengine community stronger.