Registration with webrtc server?

[ghenry]

Hi,

Is there anyway to call another p2p user without knowing their address? For example, one peer registers with the webrtc server and calls an alias and then the server returns the right address? Similar to a SIP registration?

Or is this just pure p2p? How do you recommend handling this list of online peers?

Thanks.

[cloudwebrtc]

@ghenry There is a simple signaling server, similar to the sip registration and call process, through a random ID.

flutter-webrtc-server

Usually, webrtc needs a signaling server as the sdp and candidate exchange. I think pure p2p is not easy to implement. you may need a DHT stack to exchange SDP and candidate through DHT.

[ghenry]

On Sun, 11 Aug 2019, 04:09 CloudWebRTC, ***@***.***> wrote: @ghenry <https://github.com/ghenry> There is a simple signaling server, similar to the sip registration and call process, through a random ID. flutter-webrtc-server <https://github.com/cloudwebrtc/flutter-webrtc-server>

OK, thanks. I'm currently evaluatung the effort to use both with our own SIP platform / custom setup versus writing dart-sip-ua and using that for signalling with flutter-webrtc Usually, webrtc needs a signaling server as the sdp and candidate exchange.

I think pure p2p is not easy to implement. you may need a DHT stack to exchange SDP and candidate through DHT.

So bottom line is a server or muc is needed? Thanks.

…

[ghenry]

There's also the javascript dart Bridge and the option to try JsSIP. I wonder how that would go. Or using a java or kotlin sip plugin. Have you done any work like this already? Thanks.

[cloudwebrtc]

Developing a flutter plugin using the native sip stack can be costly, and developing on every platform [android/ios/desktop] would be terrible.
In fact, I prefer to port jssip to the dart version as a pure dart sip stack. I will create a dart sip repository so that more people can participate.

[ghenry]

That was my next idea; to port JsSIP to dart and call it dart-sip-ua, but I wanted to check everything you'd done first as that will be a big job too. I presume using it via dart JS loader would be a poor idea?

[cloudwebrtc]

I presume using it via dart JS loader would be a poor idea?

I haven't test dart JS loader, so I don't know it. :(

That was my next idea; to port JsSIP to dart and call it dart-sip-ua, but I wanted to check everything you'd done first as that will be a big job too.

Before I ported the dart-sdp-transform from clux/sdp-transform, the hardest part is the grammar parser, and the other is the conversion of asynchronous functions, use Completer to replace Promise, async/await, etc.

I think the following parts should be done:

1, Necessary dependency package:
We already have some dart package dependencies.

• flutter-webrtc
• dart-events
• dart-sdp-transform
• dart-debug? Maybe only needs a logger.dart to print logs?

2，Convert *.js to *.dart.
during the time I saw JsSIP and SIP.js, it seems that JsSIP is easier to port.

3，Develop a beautiful sip client using Flutter.

[ghenry]

I presume using it via dart JS loader would be a poor idea?

I haven't test dart JS loader, so I don't know it. :(

OK, let's rule that out then.

That was my next idea; to port JsSIP to dart and call it dart-sip-ua, but I wanted to check everything you'd done first as that will be a big job too.

Before I ported the dart-sdp-transform from clux/sdp-transform, the hardest part is the grammar parser, and the other is the conversion of asynchronous functions, use Completer to replace Promise, async/await, etc.

So that's the SDP task solved already? Well, you need SDP for WebRTC anyway, so that must mean yes.

I think the following parts should be done:

1, Necessary dependency package:
We already have some dart package dependencies.

* [x]  [flutter-webrtc](https://github.com/cloudwebrtc/flutter-webrtc)

* [x]  [dart-events](https://github.com/cloudwebrtc/dart-events)

* [x]  [dart-sdp-transform](https://github.com/cloudwebrtc/dart-sdp-transform)

* [ ]  dart-debug? Maybe only needs a `logger.dart` to print logs?

Should dart-sip-ua be standalone or should it require flutter-webrtc? For my use-case I'd want all config to be able to be passed in as I plan to provision it via FCM data.

2，Convert *.js to *.dart.
during the time I saw JsSIP and SIP.js, it seems that JsSIP is easier to port.

Don't bother with SIP.js. JsSIP is older, more mature and proper. SIP.js is a fork of onSIP. Please go with JsSIP.

3，Develop a beautiful sip client using Flutter.

A simple one, but with the option to be complicated :-)

I'm happy to sponsor some of this work if you don't have an immediate requirement for it?

[cloudwebrtc]

Should dart-sip-ua be standalone or should it require flutter-webrtc?

If dart-sip-ua needs to be a client with audio/video, flutter-webrtc is necessary unless you want to use a different media engine, such as your own non-webrtc(For example, a media engine without ice/rtcp-fb.) media engine plugin, you need to wrap the sendOffer/sendAnswer/onOffer/onAnswer series of events and methods into an interface to dock your media engine.

For my use-case I'd want all config to be able to be passed in as I plan to provision it via FCM data.

If you mean use FCM data to configuration the sip account, audio/video codec order, I think it is possible. Take JsSIP as an example. When creating UA, a configuration parameter will be passed to the constructor.

// Create our JsSIP instance and run it:

Var socket = new JsSIP.WebSocketInterface('wss://sip.myhost.com');
Var configuration = {
   Sockets : [ socket ],
   Uri : 'sip:alice@example.com',
   Password : 'superpassword'
};

Var ua = new JsSIP.UA(configuration);

So that's the SDP task solved already? Well, you need SDP for WebRTC anyway, so that must mean yes.

Yes, the SDP task has been resolved.

I'm happy to sponsor some of this work if you don't have an immediate requirement for it?

I haven't got any sponsors before. If you are willing to sponsor, this will be my first time. :)
Based on my previous experience, I expect it to take around 1~1.5 months.

[ghenry]

Should dart-sip-ua be standalone or should it require flutter-webrtc?

If dart-sip-ua needs to be a client with audio/video, flutter-webrtc is necessary unless you want to use a different media engine, such as your own non-webrtc(For example, a media engine without ice/rtcp-fb.) media engine plugin, you need to wrap the sendOffer/sendAnswer/onOffer/onAnswer series of events and methods into an interface to dock your media engine.

Ah, of course. I forgot about all the native Android (java) and iOS (Objective-C) code in flutter-webrtc.

For my use-case I'd want all config to be able to be passed in as I plan to provision it via FCM data.

If you mean use FCM data to configuration the sip account, audio/video codec order, I think it is possible. Take JsSIP as an example. When creating UA, a configuration parameter will be passed to the constructor.

// Create our JsSIP instance and run it:

Var socket = new JsSIP.WebSocketInterface('wss://sip.myhost.com');
Var configuration = {
   Sockets : [ socket ],
   Uri : 'sip:alice@example.com',
   Password : 'superpassword'
};

Var ua = new JsSIP.UA(configuration);

Yep, I see that. I've used JsSIP before. Thanks.

So that's the SDP task solved already? Well, you need SDP for WebRTC anyway, so that must mean yes.

Yes, the SDP task has been resolved.

Great.

I'm happy to sponsor some of this work if you don't have an immediate requirement for it?

I haven't got any sponsors before. If you are willing to sponsor, this will be my first time. :)
Based on my previous experience, I expect it to take around 1~1.5 months.

I've sent you an email to discuss further. I'm busy with other things but will help with this project when I can anyway.

Thanks.

[joeblew99]

Hey all,

The pion stack does most of this today i think ?
pion is a webrtc stack.
I regularly use it for session init using TURN and STUN servers.
https://github.com/pion/turn

Or maybe i am missing the intent ?

[ghenry]

Thanks. So once you figure out have you can be reached through various levels of NAT etc, what do you do with that public uri? Gavin.

[ghenry]

https://janus.conf.meetecho.com/siptest.html This is to achieve something like this. Saving the parties details so you can call them from another location. How do you do this without sip?

[stale]

This issue has been automatically marked as stale because it has not had recent activity. It will be closed if no further activity occurs. Thank you for your contributions.