Add support for RTP input in srt-live-transmit #2848
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maxsharabayko
added
Type: Enhancement
|
Could you please also update the srt-live-transmit.md document with the new media type and its specifics like not actually parsing RTP you described here? |
Addressed via 3075b8e |
ethouris
reviewed
Jan 9, 2024
Codecov ReportAll modified and coverable lines are covered by tests ✅
❗ Your organization needs to install the Codecov GitHub app to enable full functionality. Additional details and impacted files@@ Coverage Diff @@
## master #2848 +/- ##
==========================================
- Coverage 67.04% 67.02% -0.02%
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Files 102 102
Lines 20368 20368
==========================================
- Hits 13656 13652 -4
- Misses 6712 6716 +4 ☔ View full report in Codecov by Sentry. |
ethouris
reviewed
Jan 10, 2024
ethouris
reviewed
Jan 10, 2024
…faults to 12 when using the RTP medium
ethouris
reviewed
Jan 10, 2024
|
Sorry that I have spread the review comments so long in time. It should be everything this time. |
maxsharabayko
approved these changes
Jan 12, 2024
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Introduce a new Source type that considers RTP separately to UDP.
The intention is to drop bytes from the head of the input buffer. This enables the RTP header to be dropped for use cases where the receiving end cannot accept RTP. The default behaviour is to skip the first 12 bytes. Where
rtpheadersizeis set, the number of bytes to skip can be varied, to deal with extended RTP headers for example.No effort is made in this initial implementation to attempt to parse the RTP headers in any way eg for reordering, extracting timing, detecting length.
This goes some way to addressing #136.