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25 Commits

Author SHA1 Message Date
Kunal
4f1d46fe0d Improvement to rtp-to-webrtc example 2023-06-07 15:06:16 -04:00
Steffen Vogel
683fc837d0 Make repo REUSE compliant 2023-05-05 11:58:49 -04:00
Pion
308f8616a3 Update CI configs to v0.10.6 
Update lint scripts and CI configs.
 2023-04-08 14:24:19 -04:00
Sean DuBois
157220e800 Run gofmt to add new build constraints 
Also remove some 1.13 specific WASM code
 2022-01-17 22:36:01 -05:00
Sean DuBois
19b78a0953 Fix typo in rtp-to-webrtc README 
VP8 -> H264
 2021-11-30 10:10:39 -05:00
Sean DuBois
4785c30a2a Add H264 instructions to rtp-to-webrtc 
Resolves #2021
 2021-11-15 13:00:05 -05:00
Andrew N. Shalaev
13c9fd5e23 Missed second slash in schema protocol 
Should be `rtp://`
 2021-10-25 07:25:20 +02:00
Kevin Staunton-Lambert
39a10ae662 Improve rtp-to-webrtc documentation 
Add quoting to the sample ffmpeg command to make it more copyable
 2021-10-24 13:17:56 -04:00
Antoine Baché
7e049ec5ec Update examples 
TestNonFatalRead now has an timeout.
Examples now use Mime types, instead of raw strings.

Fixes #839
 2021-07-02 11:49:55 -04:00
Sean DuBois
dab8a4a104 Handle errors properly in mux readLoop 
Before io.ErrShortBuffer and packetio.ErrTimeout would incorrectly end
the read loop. Now they are only printed.

Resolves #1720
 2021-03-22 11:16:11 -07:00
Nam V. Do
4942778101 Fix typo in examples 
retuned -> returned
 2021-03-16 09:44:42 -07:00
Sean DuBois
cddf631a7f Fix rtp-to-web README 
Linked to invalid commit SHA
 2021-01-29 23:13:31 -08:00
Sean DuBois
daf27bd059 Remove SampleBuilder from rtp-to-webrtc 
SampleBuilder isn't able to properly handle H264. We have multiple
issues and until resolved we shouldn't suggest it.

Relates to #1652
 2021-01-29 23:10:04 -08:00
Sean DuBois
bd5d8dea04 Simplify rtp-to-webrtc 
Remove 'Waiting for RTP Packets'. This step was needed for
PayloadType/SSRC discovery before /v3. During the migration we forgot to
remove it.
 2021-01-26 09:24:55 -08:00
Sean DuBois
9439d820c5 Don't blindly forward RTP Packets in rtp-to-webrtc 
ffmpeg produces packets that cause issues in Chromium. Instead of
validating/sanitizing just create a new packet.

Resolves #1514
 2021-01-13 09:42:28 -08:00
tarrencev
a54b74cdb7 Update pion/interceptor for NACKs 
Generate + Respond interceptors
 2020-12-14 21:40:09 -08:00
Sean DuBois
9715626a0c Revert "Read/Write RTP/RTCP packets with context" 
This change caused a ~24% performance decrease

Relates to pion/webrtc#1564

This reverts commit 47a7a64898.
 2020-12-02 20:11:06 -08:00
Atsushi Watanabe
47a7a64898 Read/Write RTP/RTCP packets with context 
Control cancel/timeout by context.
 2020-12-01 11:08:48 +09:00
Pion
a737595534 Update CI configs to v0.4.15 
Update lint scripts and CI configs.
 2020-11-16 12:18:44 -08:00
Sean DuBois
7edfb701e0 New Track API 
The Pion WebRTC API has been dramatically redesigned. The design docs
are located here [0]

You can also read the release notes [1] on how to migrate your
application.

[0] https://github.com/pion/webrtc-v3-design
[1] https://github.com/pion/webrtc/wiki/Release-WebRTC@v3.0.0
 2020-11-15 09:20:47 -08:00
aler9
cdc726201b Update examples/rtp-to-webrtc README 
Simplify gstreamer pipeline
 2020-08-20 20:35:06 -07:00
Nick Mykins
a1bbddd0d8 Improve docs around Go Modules 
adds export `GO111MODULE=on` to examples READMEs when appropriate
 2020-06-30 20:14:23 -07:00
Sean DuBois
bb3aa9717f Move to pion/ice@v2 
Removed support for trickle ice

Resolves #1274
 2020-06-28 00:01:47 -07:00
Sean DuBois
89d7de1787 Start /v3 
See #9 for the features we have planned, and the breaking changes that
may occur.
 2020-06-25 09:45:27 -07:00
a-wing
c0032c4d18 Add new example rtp-to-webrtc 
This example consumes RTP via a listening UDP socket and then sends it a
WebRTC peer
 2020-04-26 18:11:19 -07:00