Commit graph

23 commits

Author SHA1 Message Date
Daniel Gultsch 4e8ceadfbf prepare JingleRtpConnection for content-adds 2022-11-28 08:59:23 +01:00
Daniel Gultsch 304205b2e3 take senders attr into account when converting to and from sdp 2022-11-20 17:00:40 +01:00
Daniel Gultsch 70b5d8d81a set proper peer dtls setup on ice restart received 2021-11-15 21:49:31 +01:00
Daniel Gultsch 0717f9ba18 upgrade libwebrtc to m90 and enable extmap-allow-mixed 2021-05-03 09:48:46 +02:00
Daniel Gultsch 8ac97b0027 disable extmap_allow_mixed by default 2021-03-21 19:40:52 +01:00
Daniel Gultsch 6f1b71970d parse extmap-allow-mixed 2021-03-16 18:52:38 +01:00
Christopher Vollick ef24d2050b Remove Renomination from WebRTC Options
This is a feature of WebRTC that's [not standardized][1] and only
supported by libwebrtc. Since there's no support in jingle for passing
this capability from one peer to another, we're currently hard-coding
this option into both the local candidate and also the remote candidate
so they can use it.

But I'm trying to call a user that isn't using WebRTC, and renomination
is causing the call to stay in "connecting..." state for 10 or 20
seconds, sometimes longer, while both sides wait for the other to
nominate something based on their individual beliefs about the standards
they're using.

Removing this seems to make connecting relatively instantaneous.

If we want to reintroduce this feature, we should probably make a XEP so
the peers can negotiate honestly about it, and only use it if both sides
truely support the feature.

[1]: https://datatracker.ietf.org/doc/html/draft-thatcher-ice-renomination-01
2021-03-04 08:26:52 +00:00
Daniel Gultsch 8ce7bfb95e automated code clean up 2021-01-23 09:25:34 +01:00
Daniel Gultsch 2bec5459c5 properly null check ufrag and pwd before whitespace checking. fixes #3956 2021-01-03 16:05:17 +01:00
Daniel Gultsch 7bcb29c482 be more liberal in 0167 payload-type parameter parsing
some implementations will transform the following SDP coming from Firefox

m=audio 12346 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

to

<payload-type channels="1" name="telephone-event" clockrate="8000" id="101">
  <parameter value="0-15" xmlns="urn:xmpp:jingle:apps:rtp:1"/>
</payload-type>

While a missing name attribute is not legal according to the XEP; and 0-15 are
technically not just one value the following commit will accept it if there is
just one paramater.
2020-06-17 21:15:12 +02:00
Daniel Gultsch d5e3d13158 do not just assume rtcp-mux 2020-04-21 09:11:17 +02:00
Daniel Gultsch 1cc0dfad84 move sdp logging to different tag 2020-04-20 15:57:31 +02:00
Daniel Gultsch 493ca68464 add <rtcp-mux/> in description 2020-04-20 15:57:30 +02:00
Daniel Gultsch 82f9a77777 be more conservative when parsing rtp content 2020-04-20 15:57:30 +02:00
Daniel Gultsch 2ba84bd32e no need to be careful about Int parsing in session description; just fail 2020-04-20 15:57:30 +02:00
Daniel Gultsch 0bf991d95c make jingle->sdp parsing fail on some obvious errors 2020-04-20 15:57:30 +02:00
Daniel Gultsch 885ec0febe a couple of bug fixes for SessionDescription.toString() 2020-04-20 15:57:30 +02:00
Daniel Gultsch f264ef9f8b create sdp string and set on peer connection 2020-04-20 15:57:30 +02:00
Daniel Gultsch b1c0e93b34 rudimentary rtpmap to session converter 2020-04-20 15:57:30 +02:00
Daniel Gultsch b44a3aeac6 parse sdp to jingle (yet w/o transport) 2020-04-20 15:57:30 +02:00
Daniel Gultsch 18059345c8 payload-type and rtp-hdrext sdp parsing 2020-04-20 15:57:30 +02:00
Daniel Gultsch 5b1d86d67e dummy code to get sdp out of (non-working) libwebrtc 2020-04-20 15:57:30 +02:00
Daniel Gultsch e2e4390d51 untested sdp parser 2020-04-20 15:57:30 +02:00