Commit graph

31 commits

Author SHA1 Message Date
Matthew Waters
a836bd4766 webrtcbin: advertise harder the rtcp-mux-only requirement
And ignore rtcp ICE candidates

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2239>
2021-05-19 04:42:56 +00:00
Ilya Kreymer
92626535c7 webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
2021-03-01 14:42:17 +00:00
Olivier Crête
3a3965e5cf webrtc ice: Only ever request one component, it's always rtcpmux
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
2021-03-01 14:42:16 +00:00
Matthew Waters
b6038523c1 webrtcbin: use regular ice nomination by default
1. We don't currently deal with an a=ice-options in the SDP which means
   we currently violate https://tools.ietf.org/html/rfc5245#section-8.1.1
   which states: "If its peer is using ICE options (present in
   an ice-options attribute from the peer) that the agent does not
   understand, the agent MUST use a regular nomination algorithm."
2. The recommendation is default to regular nomination in both RFC5245
   and RFC8445.  libnice change for this is
   https://gitlab.freedesktop.org/libnice/libnice/-/merge_requests/125
   which requires an API break in libnice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2031>
2021-03-01 10:00:06 +00:00
Olivier Crête
df8d29e9c3 webrtcbin: Remove remnant of non-rtcp-mux mode
There was some code left that wasn't used anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1930>
2021-01-06 23:02:37 +00:00
Olivier Crête
80a56c25a6 webrtc: Set the DSCP markings based on the priority
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-10-30 16:24:40 -04:00
Sebastian Dröge
b565a7ef66 Revert "webrtc: Set the DSCP markings based on the priority"
This reverts commit 8ba08598bb.

It breaks the CI until the C# bindings are fixed.
2020-10-08 18:53:00 +03:00
Olivier Crête
8ba08598bb webrtc: Set the DSCP markings based on the priority
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
2020-10-06 16:49:08 -04:00
Olivier Crête
825a79f01f webrtcbin: Accept end-of-candidate pass it to libnice
libnice now supports the concept of end-of-candidate, so use the API
for it. This also means that if you don't do that, the webrtcbin will
never declared the connection as failed.

This requires bumping the dependency to libnice 0.1.16

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1139>
2020-09-18 18:40:58 -04:00
Nirbheek Chauhan
16d84a2816 webrtc: Clean up the userinfo unescaping code
Continuation from 04fd705906. This is
easier to understand and also avoids two copies.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1547>
2020-08-30 09:53:42 +00:00
trilene
04fd705906 webrtc: Unescape turnserver user and password
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1530>
2020-08-26 23:37:17 +01:00
Matthew Waters
9011539940 webrtc/ice: resolve .local candidates internally
Requires the system's DNS resolver to support mdns resolution.

Fixes interoperablity with recent versions of chrome/firefox that
produce .local address in for local candidates.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1139
2020-08-20 13:01:17 +10:00
Nirbheek Chauhan
d4ca8820e7 webrtc, rtmp2: Warn if the user or password aren't escaped
If the user/pass aren't escaped, the userinfo will be ambiguous and we
won't know where to split. We will accidentally get it right if the :
belongs in the password.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1481>
2020-08-03 18:12:50 +00:00
Nirbheek Chauhan
827afa206d webrtc, rtmp2: Fix parsing of userinfo in URI strings
While parsing the string, `gst_uri_from_string()` also unescapes the
userinfo. This is bad if your username contains a `:` character, since
we will then split the userinfo at the wrong location when parsing it.

To fix this, we can use the new `gst_uri_from_string_escaped()` API
that was added in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/583

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/831

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1481>
2020-08-03 18:12:50 +00:00
Matthew Waters
597c1b4ec6 webrtc: remove private properties/signals from the now public ice object
We don't want to expose all of the webrtcbin internals to the world.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-20 15:56:20 +10:00
Thibault Saunier
d9ffa3b3b2 doc: Fix spelling of GstWebRTCICE 2020-06-04 13:33:16 -04:00
Chris Ayoup
9937101e51 webrtc: move filtering properties to webrtcice
We want webrtcbin to only expose properties that are defined in JSEP, so
these additional properties should be moved out.  In order to access
them, the webrtcice instance is exposed from webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
ca754245e9 webrtc: allow setting local IP addresses
If a local IP address is specified, ICE gathering can be much faster
in environments where there are multiple IP addreses but only some are
usable (for example, if you are running docker on the machine).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
3fc8818824 webrtc: Allow toggling TCP and UDP candidates
Add some properties to allow TCP and UDP candidates to be toggled.  This
is useful in cases where someone is using this element in an environment
where it is known in advance whether a given transport will work or not
and will prevent wasting time generating and checking candidate pairs
that will not succeed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Matthew Waters
1f395e3ddb webrtc: name threads based on the element name
Makes debugging a busy loop possibly easier

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Niels De Graef
d8f61515d8 Don't pass default GLib marshallers for signals
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.

Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
2019-11-06 14:27:46 +00:00
Ilya Smelykh
e898f1565d webrtcbin: fix GInetAddress leak 2019-07-29 15:55:36 +07:00
Ali Yousuf
69e06ced7d webrtc: Fix log when adding stun server 2019-06-04 07:54:25 +00:00
Thibault Saunier
7fe3f36ac8 Minor documentation fixes 2019-05-13 11:36:27 -04:00
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
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
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
Mathieu Duponchelle
45fe050286 webrtcice: do not run host resolution from applictation thread
g_resolver_lookup_by_name is a blocking call, and should not
be run when the user sets or adds a turn-server.

https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-09-19 16:17:24 +02:00
Mathieu Duponchelle
1d6160d59c webrtcbin: New add-turn-server API
It is possible and often desirable to pass multiple ICE relays
to libnice agents, the "turn-server" property, while convenient
to use from the command line, does not allow that.

This adds a new action signal, "add-turn-server" to address that.

https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-09-19 16:17:24 +02:00
Tim-Philipp Müller
6f46792f0f webrtc: Update for g_type_class_add_private() deprecation in recent GLib 2018-06-24 00:17:26 +02:00
Matthew Waters
1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523
2018-02-02 15:02:21 +11:00