In a Real-time Transport Protocol (RTP) session the SSRC assist in determining the source endpoint, typically when an endpoint sends multiple media streams that need to be synchronized(e.g., audio and video lip-sync). SSRC identifies the synchronization source (simply called "the source"). This identifier is chosen randomly with the intent that it is unique among all the sources of the same RTP session, and if a source change its source transport address, it must also choose a new SSRC identifier.
Articles in this section
- What can I learn from the objective quality values?
- What is objective quality and how is it measured?
- What are the transport metrics shown for a conference call on the dashboard?
- What does SSRC (Synchronization Source) mean?
- What does MSID in SDP mean?
- What metrics are returned in the conference summary notification?
- Is there a way to test end users' network quality?
- How do you determine when a user drops out / rejoins a conference call?
- Can I find out which conference calls are peer to peer and which ones go through a TURN server (i.e. are relayed)?
- What metrics does callstats.io collect?