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192 commits

Author SHA1 Message Date
Olivier Crête
0e2d128bec webrtcbin: Make request-pad validation an early return 
This reduces the indendation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:07 -04:00
Olivier Crête
0f758a1730 webrtcbin: Add document for webrtcbin itself to generated doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:07 -04:00
Olivier Crête
3be72a6c86 webrtc: Reset received_caps when releasing pad 
This is to work around a race where the pad is accessed in the
webrtc main thread while being released.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:07 -04:00
Olivier Crête
b6114a7fed webrtcbin: Split pad name from mline 
The simple case where this breaks is if you add a
datachannel and want to add a new pad (a new media) after). Another
case where this is broken is if the order of the media is forced to
something different by the peer.

It's more simple to just split both things completely. In practice, the
pads will be named in the order in which they are allocated, so it
shouldn't change the current behaviour, just enable new ones.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:06 -04:00
Matthew Waters
2bed220771 webrtc: don't generate duplicate rtx payloads when bundle-policy is set 
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.

Fixes

m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
 2021-03-09 02:22:35 +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
Mathieu Duponchelle
86c009e7aa webrtc: expose transport property on sender and receiver 
As advised by !1366#note_629558 , the nice transport should be
accessed through:

> transceiver->sender/receiver->transport/rtcp_transport->icetransport

All the objects on the path can be accessed through properties
except sender/receiver->transport. This patch addresses that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1952>
 2021-01-13 19:22:42 +00:00
Mathieu Duponchelle
88e007fb21 webrtcbin: try harder not to pick duplicate media ids 
On renegotiation, or when the user has specified a mid for
a transceiver, we need to avoid picking a duplicate mid for
a transceiver that doesn't yet have one.

Also assign the mid we created to the transceiver, that doesn't
fix a specific bug but seems to make sense to me.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1902>
 2021-01-08 20:22:57 +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
51ef4557b5 webrtcstats: PLI/FIR/NACK direction are the opposite of the media 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
 2020-12-29 15:07:03 -05:00
Olivier Crête
a801018ef1 webrtc: Make ssrc map into separate data structures 
They now contain a weak reference and that could be freed later
causing strange crashes as GWeakRef are not movable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
1deb034e3d webrtcstats: Get the remote-inbound stats from the right RTPSource 
This also means that we need to get the clock-rate from the codec instead
of from the RTPSource, as the remote one doesn't include a clock rate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
1c1661b54f webrtcbin: Implement getting stats for a specific pad 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
23ea950351 webrtcstats: Also return the raw rtpsource stats for more information 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
b895240241 webrtcstats: Avoid copy of GstStructure 
Instead transfer the ownership to the new structure

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
a46c6e3a97 webrtcstats: Remove receiver side when sending 
Those are just invalid and just reflect what we sent. We'd need to parse the
RTCP XR packets from the other side to know more about those.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
ba0dfa52d2 webrtcstats: Extract statistics from the rtpjitterbuffer 
And expose them as standardised webrtc statistics

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
fc0f6db856 webrtcbin: Store the rtpjitterbuffer instances to extract stats from them 
Store them as web refs to avoid having to worry about freeing later and because
the new-jitterbuffer is on a different thread

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
d9d7814182 webrtcstats: Document all RTP missing fields according to the latest spec 
Just document all the missing fields and document which ones will never
be implemented because they depend on the codec or depayloader

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
895ea210c2 webrtcstats: RTCP computed RTT is only available at sender 
The receiver doesn't have the information to compute it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
a5c3331197 webrtcstats: Remove redundant lines 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
5d5417f271 webrtc: Remove non rtcp-mux code 
RTCP mux is now always required by the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
 2020-11-24 01:59:55 +00:00
Raul Tambre
6d300ce785 webrtc: Update libnice version requirement to 0.1.17 
Since !1366 nice_agent_get_sockets() is used, which requires 0.1.17.
Update the version requirement accordingly.

Fixes #1459.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1792>
 2020-11-11 13:41:59 +02:00
Olivier Crête
da2bd55177 webrtc: Add properties to change the socket buffer sizes to ice object 
libnice doesn't touch the kernel buffer sizes. When dealing with RTP data,
it's generally advisable to increase them to avoid dropping packets locally.
This is especially important when running multiple higher bitrate streams at
the same time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1366>
 2020-11-03 22:07:53 +00:00
Jan Schmidt
af90778314 webrtc: Fix a race on shutdown. 
The main context can disappear in gst_webrtc_bin_enqueue_task()
between checking the is_closed flag and enqueueing a source on the
main context. Protect the main context with the object lock instead
of the PC lock, and hold a ref briefly to make sure it stays alive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +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
Olivier Crête
0fbbdc5734 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Olivier Crête
7be09a5f22 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Olivier Crête
e172ca5be1 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Sebastian Dröge
cc7e98816f Revert "webrtc: Save the media kind in the transceiver" 
This reverts commit f54d8e9945.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:12 +03:00
Sebastian Dröge
849839ba97 Revert "rtptransceiver: Store the SSRC of the current stream" 
This reverts commit d1da271f25.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:07 +03:00
Sebastian Dröge
e65a8cbcf1 Revert "webrtcbin: Remove unused function" 
This reverts commit 39723dbe93.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:04 +03:00
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
39723dbe93 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
d1da271f25 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
f54d8e9945 webrtc: Save the media kind in the transceiver 
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
Olivier Crête
63f06d16db webrtcbin: Merge the RTX SSRCs from all transceivers when bundling 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1545>
 2020-09-18 14:20:03 +00:00
Matthew Waters
e2d88f0569 webrtc: propagate more errors through the promise 
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
 2020-09-14 04:04:29 +00:00
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
e15a8fcbdd webrtc/datachannel: clear the error after use 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
7489addc0a webrtc/datachannel: free previous protocol/label fields 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
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