communication protocol_error ai_generated partial

RTP SSRC collision detected: terminating stream

ID: communication/rtp-ssrc-collision-detected

Also available as: JSON · Markdown · 中文
76%Fix Rate
82%Confidence
1Evidence
2024-04-10First Seen

Version Compatibility

VersionStatusIntroducedDeprecatedNotes
FFmpeg 6.1 active
GStreamer 1.22.0 active
WebRTC (libwebrtc M120) active
Janus 1.2.0 active
MediaSoup 3.10.0 active

Root Cause

Two or more RTP participants in the same session are using the same Synchronization Source (SSRC) identifier, causing the RTP stack to terminate one stream to resolve the collision as per RFC 3550 section 8.1.

generic

中文

同一RTP会话中的两个或多个参与者使用了相同的同步源(SSRC)标识符,导致RTP堆栈终止其中一个流以解决冲突,如RFC 3550第8.1节所述。

Official Documentation

https://www.rfc-editor.org/rfc/rfc3550#section-8.1

Workarounds

  1. 85% success Ensure each RTP participant generates a random SSRC using a cryptographically secure random number generator: in C with GStreamer, set 'gst_rtp_buffer_set_ssrc(buffer, g_random_int())'.
    Ensure each RTP participant generates a random SSRC using a cryptographically secure random number generator: in C with GStreamer, set 'gst_rtp_buffer_set_ssrc(buffer, g_random_int())'.
  2. 80% success Use RTCP to detect and resolve collisions automatically: enable RTCP support in both sender and receiver, and implement the collision resolution logic from RFC 3550 section 8.2 (send BYE and re-randomize SSRC).
    Use RTCP to detect and resolve collisions automatically: enable RTCP support in both sender and receiver, and implement the collision resolution logic from RFC 3550 section 8.2 (send BYE and re-randomize SSRC).

中文步骤

  1. Ensure each RTP participant generates a random SSRC using a cryptographically secure random number generator: in C with GStreamer, set 'gst_rtp_buffer_set_ssrc(buffer, g_random_int())'.
  2. Use RTCP to detect and resolve collisions automatically: enable RTCP support in both sender and receiver, and implement the collision resolution logic from RFC 3550 section 8.2 (send BYE and re-randomize SSRC).

Dead Ends

Common approaches that don't work:

  1. Manually assign a fixed SSRC value in the RTP sender configuration 95% fail

    Static SSRC assignment guarantees collisions if multiple senders use the same value; RFC requires SSRC to be randomly chosen to minimize collision probability.

  2. Disable SSRC collision detection in the RTP stack 90% fail

    Disabling collision detection violates RFC 3550 and can cause undetected stream corruption, packet loss, and audio/video desynchronization.

  3. Increase the RTP buffer size to absorb the collision 85% fail

    Buffer size does not affect SSRC collisions; it only affects jitter handling and packet reordering.