# 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

# Contribution

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