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Alessandro Ros 4f3337f56c set SSRC of outgoing packets (#803) 
In client and server, each format now has a fixed, unique, known in
advance SSRC, that is applied to outgoing packets belonging to each
format.

This is needed to support SRTP/MIKEY, that require each format to have
a fixed, unique, and known in advance SSRC.

A secondary effect is that SETUP responses now always contain SSRCs of
each format, regardless of the fact that the first packet has been
produced or not (previously we needed at least one packet, from which
the SSRC was extracted).
2025-07-05 11:08:57 +02:00
.github
bump golangci-lint to 2.2.1 (#813)
2025-06-30 09:45:17 +02:00
examples
set default max payload size to 1450 (#811)
2025-06-29 12:51:54 +02:00
internal/teste2e
expose back rtcpreceiver, rtcpsender, rtpreorderer, rtplossdetector (#755)
2025-04-12 10:10:05 +02:00
pkg
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
scripts
remove unused scripts (#814)
2025-06-30 09:42:17 +02:00
.dockerignore
initial commit
2020-01-20 10:15:24 +01:00
.gitignore
add codecov
2021-02-13 20:13:19 +01:00
.golangci.yml
bump golangci-lint (#784)
2025-05-23 14:51:33 +02:00
async_processor_test.go
fix various race conditions when writing packets to closed clients or server sessions (#684)
2025-01-19 12:07:59 +01:00
async_processor.go
fix various race conditions when writing packets to closed clients or server sessions (#684)
2025-01-19 12:07:59 +01:00
client_format.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
client_media.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
client_play_test.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
client_reader.go
close connections in case of write errors (#613) (#655)
2024-12-14 13:45:11 +01:00
client_record_test.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
client_stats.go
add statistics to Client, ServerSession, ServerConn, ServerStream (#556) (#656)
2024-12-25 13:30:08 +01:00
client_test.go
rename auth.Validate into auth.Verify (#698)
2025-02-14 11:34:55 +01:00
client_udp_listener.go
fix various race conditions when writing packets to closed clients or server sessions (#684)
2025-01-19 12:07:59 +01:00
client.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
constants.go
fix reading and writing multicast packets in case of multiple interfaces (#413)
2023-09-11 23:36:53 +02:00
empty_timer.go
normalize file names (#173)
2023-01-22 11:44:20 +01:00
go.mod
Bump github.com/pion/rtp from 1.8.19 to 1.8.20 (#819)
2025-07-01 07:50:00 +02:00
go.sum
Bump github.com/pion/rtp from 1.8.19 to 1.8.20 (#819)
2025-07-01 07:50:00 +02:00
LICENSE
Initial commit
2020-01-20 10:08:25 +01:00
Makefile
bump golangci-lint to 2.2.1 (#813)
2025-06-30 09:45:17 +02:00
README.md
server: support reading back channels (#597) (#777)
2025-05-04 17:04:15 +02:00
restrict_network.go
do not listen on IPv6 when host is 0.0.0.0 (#240)
2023-04-10 22:42:19 +02:00
server_conn_reader.go
fix various race conditions when writing packets to closed clients or server sessions (#684)
2025-01-19 12:07:59 +01:00
server_conn.go
server: support reading back channels (#597) (#777)
2025-05-04 17:04:15 +02:00
server_handler.go
rename OnPacketLost2 into OnPacketsLost (#736)
2025-03-24 18:42:48 +01:00
server_multicast_writer.go
fix various race conditions when writing packets to closed clients or server sessions (#684)
2025-01-19 12:07:59 +01:00
server_play_test.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_record_test.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_session_format.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_session_media.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_session.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_stream_format.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_stream_media.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_stream_stats.go
server: add ServerStreamStatsFormat.LocalSSRC (#804)
2025-06-28 16:56:44 +02:00
server_stream.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_tcp_listener.go
replace new() with initialize() (#490)
2023-12-26 12:48:35 +01:00
server_test.go
set SSRC of outgoing packets (#803)
2025-07-05 11:08:57 +02:00
server_udp_listener.go
fix various race conditions when writing packets to closed clients or server sessions (#684)
2025-01-19 12:07:59 +01:00
server.go
update documentation (#788)
2025-05-29 19:40:03 +02:00
stats_conn.go
add statistics to Client, ServerSession, ServerConn, ServerStream (#556) (#656)
2024-12-25 13:30:08 +01:00
stats_session.go
rename RTPJitter into RTPPacketsJitter (#669)
2024-12-25 15:14:40 +01:00
transport_test.go
rename ClientTransport into Transport
2021-10-22 17:53:51 +02:00
transport.go
update docs
2022-05-10 10:28:52 +02:00

README.md

gortsplib

Test Lint Go Report Card CodeCov PkgGoDev

RTSP 1.0 client and server library for the Go programming language, written for MediaMTX.

Go ≥ 1.23 is required.

Features:

  • Client
    • Query servers about available media streams
    • Read media streams from a server ("play")
      • Read streams with the UDP, UDP-multicast or TCP transport protocol
      • Read TLS-encrypted streams (TCP only)
      • Switch transport protocol automatically
      • Read selected media streams
      • Pause or seek without disconnecting from the server
      • Write to ONVIF back channels
      • Get PTS (relative) timestamp of incoming packets
      • Get NTP (absolute) timestamp of incoming packets
    • Write media streams to a server ("record")
      • Write streams with the UDP or TCP transport protocol
      • Write TLS-encrypted streams (TCP only)
      • Switch transport protocol automatically
      • Pause without disconnecting from the server
  • Server
    • Handle requests from clients
    • Validate client credentials
    • Read media streams from clients ("record")
      • Read streams with the UDP or TCP transport protocol
      • Read TLS-encrypted streams (TCP only)
      • Get PTS (relative) timestamp of incoming packets
      • Get NTP (absolute) timestamp of incoming packets
    • Serve media streams to clients ("play")
      • Write streams with the UDP, UDP-multicast or TCP transport protocol
      • Write TLS-encrypted streams (TCP only)
      • Compute and provide SSRC, RTP-Info to clients
      • Read ONVIF back channels
  • Utilities
    • Parse RTSP elements
    • Encode/decode RTP packets into/from codec-specific frames

Table of contents

Examples

API Documentation

Click to open the API Documentation

RTP Payload Formats

In RTSP, media streams are transmitted by using RTP packets, which are encoded in a specific, codec-dependent, format. This library supports formats for the following codecs:

Video

codec documentation encoder and decoder available
AV1 link ✔️
VP9 link ✔️
VP8 link ✔️
H265 link ✔️
H264 link ✔️
MPEG-4 Video (H263, Xvid) link ✔️
MPEG-1/2 Video link ✔️
M-JPEG link ✔️

Audio

codec documentation encoder and decoder available
Opus link ✔️
Vorbis link
MPEG-4 Audio (AAC) link ✔️
MPEG-1/2 Audio (MP3) link ✔️
AC-3 link ✔️
Speex link
G726 link
G722 link ✔️
G711 (PCMA, PCMU) link ✔️
LPCM link ✔️

Other

codec documentation encoder and decoder available
MPEG-TS link

Specifications

name area
RFC2326, RTSP 1.0 protocol
RFC7826, RTSP 2.0 protocol
RFC8866, SDP: Session Description Protocol SDP
RTP Payload Format For AV1 (v1.0) payload formats / AV1
RTP Payload Format for VP9 Video payload formats / VP9
RFC7741, RTP Payload Format for VP8 Video payload formats / VP8
RFC7798, RTP Payload Format for High Efficiency Video Coding (HEVC) payload formats / H265
RFC6184, RTP Payload Format for H.264 Video payload formats / H264
RFC3640, RTP Payload Format for Transport of MPEG-4 Elementary Streams payload formats / MPEG-4 audio, MPEG-4 video
RFC2250, RTP Payload Format for MPEG1/MPEG2 Video payload formats / MPEG-1 video, MPEG-2 audio, MPEG-TS
RFC2435, RTP Payload Format for JPEG-compressed Video payload formats / M-JPEG
RFC7587, RTP Payload Format for the Opus Speech and Audio Codec payload formats / Opus
Multiopus in libwebrtc payload formats / Opus
RFC5215, RTP Payload Format for Vorbis Encoded Audio payload formats / Vorbis
RFC4184, RTP Payload Format for AC-3 Audio payload formats / AC-3
RFC6416, RTP Payload Format for MPEG-4 Audio/Visual Streams payload formats / MPEG-4 audio
RFC5574, RTP Payload Format for the Speex Codec payload formats / Speex
RFC3551, RTP Profile for Audio and Video Conferences with Minimal Control payload formats / G726, G722, G711, LPCM
RFC3190, RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio payload formats / LPCM
Codec specifications codecs
Golang project layout project layout

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