Commit graph

86 commits

Author SHA1 Message Date
David Gunzinger
e2e86658f2 webrtc: fix type of max-retransmits, make it work 2019-08-13 12:17:13 +02:00
Sebastian Dröge
28b0be4036 rtptransceiver: Remove direction setter and vfunc and replace it by a property
It was changed from a function to a property in the latest WebRTC spec.
2019-08-06 12:22:21 +00:00
Jakub Adam
831b124976 webrtcbin: Support data channel SDP offers from Chrome
When negotiating a data channel, Chrome as recent as 75 still uses SDP
based on version 05 of the SCTP SDP draft, for example:

 m=application 9 DTLS/SCTP 5000
 a=sctpmap:5000 webrtc-datachannel 1024

Implement support for parsing SCTP port out of SDP message with sctpmap
attribute. Fixes data channel negotiation with Chrome browser.
2019-07-29 22:04:08 +00:00
Ilya Smelykh
e898f1565d webrtcbin: fix GInetAddress leak 2019-07-29 15:55:36 +07:00
Mathieu Duponchelle
b42d98ca19 webrtcdatachannel: inherit directly from GObject
There's no reason for it to inherit from GstObject apart from
locking, which is easily replaced, and inheriting from
GInitiallyUnowned made introspection awkward and needlessly
complicated.
2019-07-16 21:35:47 +00:00
Sebastian Dröge
329b2d3a6a webrtcbin: Don't assert if an SDP media can't be converted to caps
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1008
2019-07-08 07:18:41 +00:00
Matthew Waters
cef839533e webrtcbin: use the latest self-generated SDP as the basis for renegotiations
Fixes multiple errors when a webrtcbin renegotiation can switch between the
offerer and the answerer.
2019-07-03 23:44:15 +00:00
Philippe Normand
36de11520e webrtc: Fix data-channel send-string doc 2019-06-23 17:03:32 +01:00
Mathieu Duponchelle
9023ac1c95 webrtcbin: fix DTLS when receivebin is set to DROP
Regression introduced by b4bdcf15b7

This commit prevents the handshake from reaching dtlsdec when
the receive state of the receive bin is set to DROP (for example
when transceivers are sendonly).

This preserves the intent of the commit, by blocking the bin
at its sinks until the receive state is no longer BLOCK, but
makes sure the handshake still goes through, by only dropping
data at the src pads, as was the case before.
2019-06-19 18:04:14 +00:00
Ali Yousuf
69e06ced7d webrtc: Fix log when adding stun server 2019-06-04 07:54:25 +00:00
Matthew Waters
95488812b2 webrtc: fix the location of signalling-state change notification
1. The spec indicates that the notification should occur near the end of
   'setting the description' processing
2. The current location with the drop of the lock could cause the 'check
   if negotiation is needed' logic to execute and become confused about
   the state of the webrtcbin's current local descriptions.
   In the bad case, the following assertions could be hit:
   g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
   g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));

Moving the signalling state change later in the set description task
means that checking for a renegotiation will early abort as the
signalling state is not STABLE before the session description and
transceivers have been updated.
2019-06-04 05:43:43 +00:00
Matthew Waters
f8911deccf webrtc: only set sctp ports if they are different
SCTPassociation will complain if we do that while running and resetting
is not something we support at the moment
2019-05-30 21:33:09 +10:00
Matthew Waters
979daea7f2 tests/webrtc: fix racy test with a prenegotiated data channel
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice.  Use an atomic add instead.
2019-05-30 21:33:09 +10:00
Matthew Waters
be011d2086 webrtc/dc: move some code from webrtcbin into the datachannel 2019-05-30 21:33:09 +10:00
Matthew Waters
a51db86ac4 webrtc: hold onto any unknown ICE candidates until the next SDP set
It is very possible for badly behaving signalling or peers to send
us ICE candidates before we receive an SDP.  While we had consideration
for that on the first set SDP, subsequent SDP's could result in
misconfigured ICE transports.  Expand the previous code to also take
into account reconfigurations.
2019-05-30 21:33:09 +10:00
Matthew Waters
177aa22bcd webrtc: Initial support for stream addition/removal
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
2019-05-30 21:33:09 +10:00
Matthew Waters
033e55695f webrtcbin: expose the transceiver as a pad property 2019-05-30 21:33:09 +10:00
Matthew Waters
c3c4b07ad3 webrtc/transceiver: add a set_direction function
Matches the setDirection() from the W3C spec and allows changing the
transceiver direction at the next negotiation cycle.
2019-05-30 21:33:09 +10:00
Matthew Waters
6ad0edbe92 webrtc: track and log more rtpbin state
like bye's timeouts, validation, activation, etc
2019-05-30 21:33:09 +10:00
Matthew Waters
2df7da85fe webrtc: add support for intersecting inactive transceiver directions 2019-05-30 21:33:09 +10:00
Matthew Waters
5ea7031bd0 webrtc: mark remote/local-description as readonly 2019-05-30 21:32:06 +10:00
Matthew Waters
19b3d744d8 webrtc: don't reuse stopped transceivers at all 2019-05-30 21:26:46 +10:00
Matthew Waters
4d34fe7617 webrtc: also check for a null mid to signify an unassociated transceiver
We always give our transceivers an mline on creation so that check is
not useful by itself
2019-05-30 21:26:46 +10:00
Matthew Waters
00977f263a webrtc: only check sink pads for a 'sink pads have caps' check 2019-05-30 21:26:46 +10:00
Matthew Waters
bd92b2f7c4 webrtc: fix answer creation with multiple streams and similar caps 2019-05-30 21:26:46 +10:00
Philippe Normand
9595a7a721 webrtcbin: Expose current and pending local/remote description properties
They are already handled in the property getter and setter functions but were
not formally declared in the GObject class.
2019-05-30 10:35:58 +01:00
Niels De Graef
7cd4064425 webrtc: Fix some signals' GIR annotations
This will lead to wrong bindings otherwise (and creates more correct
expectations for developers).
2019-05-17 15:28:54 +02:00
Thibault Saunier
47a49f3381 docs: Build documentation with hotdoc 2019-05-13 17:00:00 -04:00
Thibault Saunier
7fe3f36ac8 Minor documentation fixes 2019-05-13 11:36:27 -04:00
Niels De Graef
ce92cb81a0 webrtc: Fix signals documentation
Some GIR annotations were incorrect or even missing. The former isn't
good for bindings, while the latter is especially annoying for signal
handlers, as that means your arguments will get the wrong names in the
rendered documentation.
2019-05-09 14:19:01 +02:00
Mathieu Duponchelle
a2779ef366 webrtcbin: fix pt selection for FEC and RTX when BUNDLE
When we offer bundled media, payload types must be unique
across all bundled media, as they will be multiplexed in the
same session.
2019-03-15 18:37:51 +01:00
Mathieu Duponchelle
08858d753c webrtcbin: add get-transceiver signal
get-transceivers is not introspectable, and a method to get a
transceiver by index is convenient.
2019-03-12 21:04:48 +00:00
Jan Alexander Steffens (heftig)
dc0e95acab webrtcbin: Filter transport stream stats by ssrc
Since the addition of BUNDLE support, the pads and the transceivers
share a single transport stream. When getting stats from the stream,
filter by the ssrc of the current pad to avoid merging the stats for
different pads.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/889
2019-03-12 01:40:59 +00:00
Jan Alexander Steffens (heftig)
926ff109b9 webrtcbin: Syntax cleanup 2019-03-12 01:40:59 +00:00
Matthew Waters
2a1176973a webrtc: fix rtx + bundle
If bundle was used in combination with rtx, only the bundled transport
stream would have correctly configured rtx parameters.

Iterate over the payloads upfront in the bundled case to ensure the
correct payload mapping is set for the RTX elements.
2019-02-15 08:19:51 +00:00
Mathieu Duponchelle
85c75bb23b webrtc: expose ice-transport-policy property
This is the equivalent of iceTransportPolicy in the RTCConfiguration
dictionary.

Only two values are implemented:

* all: default behaviour
* relay: only gather relay candidates

The third member of the iceTransportPolicy enum, "public", is
obsolete.
2019-01-23 22:47:51 +00:00
Tim-Philipp Müller
9eb7f7cbc7 webrtc: include stdlib.h for atoi()
Fixes #857
2018-12-31 12:09:42 +00:00
Matthew Waters
b4bdcf15b7 webrtc/receive-bin: block pads before dtlssrtpdec:
Fixes SSL errors in fast-start scenarios and whenever media stream may
be received before an answer is set.
2018-12-19 00:44:06 +00:00
Matthew Waters
26a5cbddbb webrtcbin: only change the receive state after setting the dtls-client
Doing so before will cause SSL errors with fast-start implementations
like Chrome or if media data arrives before an answer.
2018-12-19 00:44:06 +00:00
Matthew Waters
0a3f662ed6 webrtc: A couple of documentation fixes
set-*-description only takes the combined GstWebRTCSessionDescription
object
2018-12-19 00:44:06 +00:00
Nirbheek Chauhan
9504fc7174 meson: Add missing gio dep to webrtcbin plugin
It's usually pulled in implicitly through gstsdp_dep, but it's
actually a private dependency there. Fixes a build failure on Windows
with newer Meson.
2018-12-05 19:58:44 +05:30
Jordan Petridis
1f562870ee Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 14:18:26 +00:00
Matthew Waters
57a006d8a5 tests/webrtc: use the existing functions in the plugin
Instead of redefining our own, use the function implementations in
webrtcsdp.c and utils.c
2018-11-26 17:13:08 +11:00
Matthew Waters
14ee6f9d35 webrtc: fix typo in RTCRemoteOutboundRTPStreamStats 2018-11-26 16:21:58 +11:00
Matthew Waters
a42fdbb012 webrtc: add a few comments on bundle and src pad exposure 2018-11-26 16:21:19 +11:00
Matthew Waters
6f91a191de webrtcbin: factor out dtls fingerprint setting 2018-11-26 16:20:02 +11:00
Matthew Waters
3a2566c61f webrtc: remove extra 'pad' from log line 2018-11-26 16:12:03 +11:00
Matthew Waters
5ecca0bb22 webrtc: move some functions to the appropriate files 2018-11-26 16:07:57 +11:00
Harshad Khedkar
9ad618e487 Webrtcbin : Need to use 'host' from gst_uri_get_host s libnice agent expects it
Currently master code of gst1-plugins-bad use plain-string host name while passing it to
libnice agent: nice_agent_set_relay_info() in gstwebrtcice.c while adding turn_server(_add_turn_server).

It is observered that if we don't convert the host parameter by using gst_uri_get_host, it fails in libnice agent(0.1.14-1).

Code does, actually, set the host correctly but while passing params to nice_agent_set_relay_info, it uses incorrect one.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/823
2018-11-22 18:47:13 +05:30
Tim-Philipp Müller
247cd2113e webrtc: update default libnice options
Uses feature options now.
2018-11-02 20:16:56 +00:00