|
1 |
| -## Overview |
2 |
| - |
3 |
| -This project aims to be a convenient location for [WebRTC](https://www.w3.org/TR/webrtc/) library developers to perform interoperability tests. |
4 |
| - |
5 |
| -## Who can Participate |
6 |
| - |
7 |
| -The project is open to everyone. |
8 |
| - |
9 |
| -The developers likely to interested are those involved in WebRTC projects. |
10 |
| - |
11 |
| -## Participating Libraries |
12 |
| - |
13 |
| -The libraries that currently have a Client and Server implementation are: |
14 |
| - |
15 |
| - - [aiortc](https://github.com/aiortc/aiortc): WebRTC and ORTC implementation for Python using asyncio. |
16 |
| - - [libdatachannel](https://github.com/paullouisageneau/libdatachannel): C/C++ WebRTC Data Channels and Media Transport standalone library (bindings for [Rust](https://github.com/lerouxrgd/datachannel-rs), [Node.js](https://github.com/murat-dogan/node-datachannel), and [Unity](https://github.com/hanseuljun/datachannel-unity)) |
17 |
| - - [Pion](https://github.com/pion/webrtc): Pure Go implementation of the WebRTC API. |
18 |
| - - [webrtc-rs](https://github.com/webrtc-rs/webrtc): A pure Rust implementation of WebRTC stack. Rewrite [Pion](https://github.com/pion/webrtc) WebRTC stack in Rust. |
19 |
| - - [SIPSorcery](https://github.com/sipsorcery-org/sipsorcery): A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps. |
20 |
| - - [werift-webrtc](https://github.com/shinyoshiaki/werift-webrtc): WebRTC Implementation for TypeScript (Node.js) |
21 |
| - |
22 |
| -Additional libraries/applications that currently have a Server implementation are: |
23 |
| - |
24 |
| - - [gstreamer](https://gstreamer.freedesktop.org/) `master` branch, commit ID `637b0d8dc25b660d3b05370e60a95249a5228a39` 20 Aug 2021 (gst-build commit ID `ebcca1e5ead27cab1eafc028332b1984c84b10b2` 26 Mar 2021). |
25 |
| - - [janus](https://janus.conf.meetecho.com/) version `0.10.7`, commit ID `04229be3eeceb28dbc57a70a57928aab223895a5`. |
26 |
| - - [kurento](https://www.kurento.org/) version `6.16.1~1.g907a859`. |
27 |
| - - [libwebrtc](https://webrtc.googlesource.com/src/) `m90` branch, commit ID `a4da76a880d31f012038ac721ac4abc7ea3ffa2d`, commit date `Fri Apr 9 21:03:39 2021 -0700`. |
28 |
| - |
29 |
| -## Interoperability Tests |
30 |
| - |
31 |
| -The current interoperability tests are: |
32 |
| - |
33 |
| - - **[Peer Connection Test](doc/PeerConnectionTestSpecification.md)**: The initial, and simplest, test is a WebRTC `Server Peer` and/or `Client Peer` that tests the ability to negotiate a peer connection up to a successful DTLS handshake. **A description of how the Peer Connection Test works is available [here](doc/PeerConnectionTestSpecification.md)**. |
34 |
| - |
35 |
| - - **[Data Channel Echo Test](doc/DataChannelEchoTestSpecification.md)**: This test builds on the [Peer Connection Test](doc/PeerConnectionTestSpecification.md) and adds a `data channel` test. It tests the ability of the peers to create a data channel and then checks that the `Server Peer` can echo a string message sent by the `Client Peer`. |
36 |
| - |
37 |
| -## Peer Connection Test Results |
38 |
| -Test run at 2024-10-12 22:03:44.834213 |
39 |
| - |
40 |
| -| Server | aiortc | libdatachannel | pion | sipsorcery | werift | |
41 |
| -|--------|--------|--------|--------|--------|--------| |
42 |
| -| aiortc | ✔ | ✔ | ✔ | | ✔ | |
43 |
| -| gstreamer | ✔ | | ✔ | ✔ | ✔ | |
44 |
| -| janus | ✔ | ✔ | ✔ | ✔ | ✔ | |
45 |
| -| kurento | ✔ | ✔ | | ✔ | ✔ | |
46 |
| -| libdatachannel | ✔ | ✔ | ✔ | ✔ | ✔ | |
47 |
| -| libwebrtc | ✔ | ✔ | ✔ | ✔ | ✔ | |
48 |
| -| pion | ✔ | ✔ | ✔ | ✔ | ✔ | |
49 |
| -| sipsorcery | ✔ | ✔ | ✔ | ✔ | ✔ | |
50 |
| -| werift | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 1 | +## Overview |
| 2 | + |
| 3 | +This project aims to be a convenient location for [WebRTC](https://www.w3.org/TR/webrtc/) library developers to perform interoperability tests. |
| 4 | + |
| 5 | +## Who can Participate |
| 6 | + |
| 7 | +The project is open to everyone. |
| 8 | + |
| 9 | +The developers likely to interested are those involved in WebRTC projects. |
| 10 | + |
| 11 | +## Participating Libraries |
| 12 | + |
| 13 | +The libraries that currently have a Client and Server implementation are: |
| 14 | + |
| 15 | + - [aiortc](https://github.com/aiortc/aiortc): WebRTC and ORTC implementation for Python using asyncio. |
| 16 | + - [libdatachannel](https://github.com/paullouisageneau/libdatachannel): C/C++ WebRTC Data Channels and Media Transport standalone library (bindings for [Rust](https://github.com/lerouxrgd/datachannel-rs), [Node.js](https://github.com/murat-dogan/node-datachannel), and [Unity](https://github.com/hanseuljun/datachannel-unity)) |
| 17 | + - [Pion](https://github.com/pion/webrtc): Pure Go implementation of the WebRTC API. |
| 18 | + - [webrtc-rs](https://github.com/webrtc-rs/webrtc): A pure Rust implementation of WebRTC stack. Rewrite [Pion](https://github.com/pion/webrtc) WebRTC stack in Rust. |
| 19 | + - [SIPSorcery](https://github.com/sipsorcery-org/sipsorcery): A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps. |
| 20 | + - [werift-webrtc](https://github.com/shinyoshiaki/werift-webrtc): WebRTC Implementation for TypeScript (Node.js) |
| 21 | + |
| 22 | +Additional libraries/applications that currently have a Server implementation are: |
| 23 | + |
| 24 | + - [gstreamer](https://gstreamer.freedesktop.org/) `master` branch, commit ID `637b0d8dc25b660d3b05370e60a95249a5228a39` 20 Aug 2021 (gst-build commit ID `ebcca1e5ead27cab1eafc028332b1984c84b10b2` 26 Mar 2021). |
| 25 | + - [janus](https://janus.conf.meetecho.com/) version `0.10.7`, commit ID `04229be3eeceb28dbc57a70a57928aab223895a5`. |
| 26 | + - [kurento](https://www.kurento.org/) version `6.16.1~1.g907a859`. |
| 27 | + - [libwebrtc](https://webrtc.googlesource.com/src/) `m90` branch, commit ID `a4da76a880d31f012038ac721ac4abc7ea3ffa2d`, commit date `Fri Apr 9 21:03:39 2021 -0700`. |
| 28 | + |
| 29 | +## Interoperability Tests |
| 30 | + |
| 31 | +The current interoperability tests are: |
| 32 | + |
| 33 | + - **[Peer Connection Test](doc/PeerConnectionTestSpecification.md)**: The initial, and simplest, test is a WebRTC `Server Peer` and/or `Client Peer` that tests the ability to negotiate a peer connection up to a successful DTLS handshake. **A description of how the Peer Connection Test works is available [here](doc/PeerConnectionTestSpecification.md)**. |
| 34 | + |
| 35 | + - **[Data Channel Echo Test](doc/DataChannelEchoTestSpecification.md)**: This test builds on the [Peer Connection Test](doc/PeerConnectionTestSpecification.md) and adds a `data channel` test. It tests the ability of the peers to create a data channel and then checks that the `Server Peer` can echo a string message sent by the `Client Peer`. |
| 36 | + |
| 37 | +## Peer Connection Test Results |
| 38 | +Test run at 2024-10-12 21:10:32.093918 |
| 39 | + |
| 40 | +| Server | aiortc | libdatachannel | pion | sipsorcery | werift | |
| 41 | +|--------|--------|--------|--------|--------|--------| |
| 42 | +| aiortc | ✔ | | ✔ | | ✔ | |
| 43 | +| gstreamer | ✔ | | ✔ | ✔ | ✔ | |
| 44 | +| janus | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 45 | +| kurento | ✔ | ✔ | | ✔ | ✔ | |
| 46 | +| libdatachannel | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 47 | +| libwebrtc | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 48 | +| pion | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 49 | +| sipsorcery | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 50 | +| werift | ✔ | ✔ | ✔ | ✔ | ✔ | |
| 51 | + |
| 52 | +## Data Channel Echo Test Results |
| 53 | +Test run at 2024-10-12 21:50:30.476743 |
| 54 | + |
| 55 | +| Server | libdatachannel | sipsorcery | werift | |
| 56 | +|--------|--------|--------|--------| |
| 57 | +| libdatachannel | ✔ | ✔ | ✔ | |
| 58 | +| sipsorcery | ✔ | ✔ | ✔ | |
| 59 | +| werift | ✔ | ✔ | ✔ | |
| 60 | +## Get Started |
| 61 | + |
| 62 | +If you are interested in adding a library to this project the recommended steps are listed below. The steps don't necessarily have to be completed in any specific order. |
| 63 | + |
| 64 | + - Write a Peer Connection Test Client application according to the [specification](doc/EchoTestSpecification.md#client-peer-operation) or base it off an [existing application](doc/EchoTestSpecification.md#view-the-code). |
| 65 | + |
| 66 | + - Test your client by building and running one of the [Peer Connection Test Servers](https://github.com/sipsorcery/webrtc-echoes/blob/master/doc/EchoTestSpecification.md#view-the-code) or you can use one of the [Peer Connection Test Server Docker Images](https://github.com/sipsorcery?tab=packages&q=webrtc): |
| 67 | + |
| 68 | +```` |
| 69 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/aiortc-webrtc-echo |
| 70 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/gstreamer-webrtc-echo |
| 71 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/janus-webrtc-echo |
| 72 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/kurento-webrtc-echo |
| 73 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/libdatachannel-webrtc-echo |
| 74 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/libwebrtc-webrtc-echo |
| 75 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/pion-webrtc-echo |
| 76 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/sipsorcery-webrtc-echo |
| 77 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/webrtc-rs-webrtc-echo |
| 78 | +docker run -it --rm --init -p 8080:8080 ghcr.io/sipsorcery/werift-webrtc-echo |
| 79 | +```` |
| 80 | + |
| 81 | +- If you encounter any problems open an [Issue](https://github.com/sipsorcery/webrtc-echoes/issues). When done submit a [Pull Request](https://github.com/sipsorcery/webrtc-echoes/pulls) for your application. |
| 82 | + |
| 83 | +- Repeat the process for a [Peer Connection Test Server](doc/PeerConnectionTestSpecification.md#server-peer-operation). |
| 84 | + |
| 85 | +- Create a [Dockerfile](doc/EchoTestDockerRequirements.md) and add a Pull Request for it so your Peer Connection Test application(s) can be included in the automated testing. |
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