Sean DuBois
d9e6b8b61f
Update examples to only generate SDP when candidates are done gathering
2018-08-11 13:56:28 -07:00
Sean DuBois
cf6e2d9e03
Enable STUN by default in examples
2018-07-21 12:27:38 -07:00
backkem
ab6910899c
api: support a custom media engine
2018-07-16 14:20:18 -07:00
backkem
f7c5ecd57f
api: fix typos
2018-07-16 14:20:18 -07:00
backkem
7f682d2d2e
api: match WebRTC api more closely
2018-07-16 14:20:18 -07:00
Sean DuBois
2564609560
Disable STUN in examples temporarily
...
When we have multiple candidates that resolve we need to share DTLS
state across them
2018-07-14 13:26:01 -07:00
Sean DuBois
f440fc32d4
Finish STUN implementation
...
* Do not increment component id (this is used for marking RTP/RTCP)
* Add STUN to all examples so that they work out of the box
* Cast Addr from STUN client to UdpAddr instead of parsing
2018-07-11 21:58:49 -07:00
Sean DuBois
76a07068c9
Use HTTPS for jsfiddle examples
2018-07-07 11:32:58 -07:00
Sean DuBois
7aa47c7d99
Add empty css files to jsfiddle demos
2018-07-07 11:32:58 -07:00
Sean DuBois
074e3391bf
Move JSfiddle snippets to git
...
jsfiddle provides a way to create snippets from Github via a URL. This
way we can still provide easy demos, but get all the nice things from
having them in Git
Closes #32
2018-07-07 11:32:58 -07:00
Raphael Randschau
98ea0b791e
fix warnings in examples
2018-07-06 15:23:40 -07:00
Raphael Randschau
4f6983307f
update examples with new constructor
2018-07-06 15:23:40 -07:00
John Bradley
cfba14cfea
Add H264 send/receive and packetization support
2018-07-05 17:16:51 -05:00
Sean DuBois
c6d8334cf7
Update gstreamer-send jsfiddle
2018-07-04 00:49:23 -07:00
Sean DuBois
24a312c34d
Fix poorly named function in RTCPeerConnection
...
CreateOffer -> CreateAnswer currently `pion-WebRTC` can only generate
offers not answers.
2018-07-04 00:46:01 -07:00
Sean DuBois
3b3ed9a544
Fix lint, vet and fmt errors
2018-07-03 21:11:25 -07:00
Sean DuBois
366f9ec268
Implement sample count generation in gstreamer-send
...
Audio+Video now works in gstreamer-send
2018-07-03 20:55:10 -07:00
Sean DuBois
a5cf1702e8
Use unique names for static globals in GStreamer cgo
2018-07-03 17:34:16 -07:00
Sean DuBois
b1da546d24
Implement multi-pipeline gstreamer-send example
2018-07-03 17:11:56 -07:00
John Bradley
912a8e18f8
Add opus sending support
2018-07-03 18:00:45 -05:00
Sean DuBois
f7ae8e3d0a
Copy @backkem doc fixes to gstreamer-receive and save-to-disk
2018-07-03 11:42:40 -07:00
backkem
8df477e38d
gstreamer-send: document running in Windows.
2018-07-03 11:18:46 -07:00
Sean DuBois
6eb22ad669
Add 'gstreamer-send' to README.md
2018-07-03 00:10:38 -07:00
Sean DuBois
093a4efac4
Gofmt
2018-07-02 21:58:08 -07:00
John R. Bradley
799e02d8f8
Add VP8 payloading
2018-07-02 23:28:53 -05:00
Sean DuBois
de2fb09778
Add support for receiving audio
...
Update gstreamer-receive to create pipelines for each input.
Currently we don't allow the user to pass in what codecs they support and we don't
take into account the offer. The API will need to be updated to catch
both these signaling errors. The user will pass a slice of support
codecs.
2018-07-01 02:04:47 -07:00
Sean DuBois
5bf9d5af34
Add ICE connection state change notification and timeouts
2018-06-30 02:57:47 -07:00
Sean DuBois
a623369bca
Fix lint errors
2018-06-24 23:11:14 -07:00
Sean DuBois
5235a4f78c
Prepare for send-peer
...
Only thing left is creating the RTP packets, using packets directly
works
2018-06-24 19:48:45 -07:00
Sean DuBois
51136804ac
Significant send progress
...
* GStreamer sends RTP packets to Go
* pion-WebRTC generates proper SDP, and has certificate ready
* Just need to implement SRTP functionality and rough MVP should be done
2018-06-22 01:04:07 -07:00
