See also Farstream/ApiProblems
I've also listed the missing requirements for Voice over LTE (VoLTE) and rtcweb.
Voice over LTE (VoLTE) has extra requirements. WebRTc
RTP plugin
Add an "extra-send-codecs" property to FsSession and add it to the "farsight-send-codec-changed" message
- RTP/RTCP multiplexing/demultiplexing (to send both on the same UDP port)
- Comfort noise (write Free software VAD and CN generator)
- Add REDundant audio data (RFC 2198)
- Support multiple stun servers in the rawudp plugin and possibly also libnice
- Document
- That you have to be playing for most stuff to happen (like getting stun replies)
- Re-write codec discovery
- Make it possible to use the missing-element message to ask the user to install extra codecs, so keep list of pay/depay without enc/dec
- Make it possible to specify input/output caps so we can send pre-encoded data, etc
- Make the codec cleaner and hopefully remove the h263/amr hacks
- SRTP: There is a branch of gst-plugins-bad that integrates libsrtp. The farstream integration needs to be done too
Tests to write
- Two sessions in the same conference
- Packets with invalid payload types
- Setting invalid payload types as local or remote codecs (in the 35-96 range).
- Errors:
- New stream with invalid participant (invalid... from another type or from
- another conf?)
- New stream with invalid participant (invalid... from another type or from
- Change codec ids while its running...
- New codec with the same PT while its running
- H263-1998 (use h263 or h263+ encoder depending on the properties)
- 3 way negotiation
- success
- failed
- Test codec cache:
- empty
- save
- load
- Test non-rtcp case (drop rtcp, filter candidate, provide wrong candidate?)
- Test lack of rtcp with more than one participant (it should fail)
- Change the udp port of a stream while it's running (rawudp transmitter, multicast transmitter?)
- Generate DTMF
- Sound (write dtmf detector?)
- Receive DTMF
Events (without dtmf detector? only check if GstMessages are received maybe?)
- Changing a running pipeline
- Add a session to a running pipeline
- Destroy a session in a running pipeline
- Re-add a destroying session in a running pipline
- Write automated tests for the live adder
- Set to playing more than once?
Other plugins
Port msnav, yahoowebcam to the new api
Generic GStreamer RTP enhancements
- Make reverse propagation of SDP optional parameters to the encoders
- Speex
- H263
- H264