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koolandrew

SIP over WebSocket (RFC 7118)

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Vodia PBX    0

Interesting... I don't think it will be difficult to get this going, especially because we are already massively using websocket for HTTP (and also the soft phone in the browser, using WebRTC).

 

But we can't do this just out of curiosity. The question is if any product requires this or there is any demand for that?

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koolandrew    0

HI,

 

That is what i was trying to accomplish. I dont know anything about this softphone using WebRTC.

 

I was thinking about using our own softphone utilizing WebRTC. What i was really trying to do, was create a full featured softphone, using WebRTC. I was trying to eliminate the installation etc part, like everyone else!!!

 

I have tried testing webrtc using your code and it really doesnt work.

 

There is little documentation, that i could find, regarding OS or browser.

 

If there are any updates on this topic, i would appreciate your response as i thought it was kind of a dead project for your development.

 

Thanks

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Vodia PBX    0

Well, we are using some super-stripped-down version of SIP. However the main problem with a 3rd party WebRTC-based phone will be that WebRTC requires the media session is set up using ICE, with testing out candidates and using DTLS.

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koolandrew    0

Please explain this to me. I dont see it anywhere on the user login. I see a call from phone, but what does that mean.

 

Could you please provide an explanation about this.

 

thanks

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Vodia PBX    0

Call from phone means that the PBX will use a SIP phone to start the call (through third party call control). Call from browser means that it will use the web browser to make the phone call; this currently works only with Chrome.

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koolandrew    0

ok thanks. I never use chrome, so i had no idea what you meant. :angry: . I tested it, and it seems to work a little bit for outbound, but i could never get it to ring on the inbound.

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koolandrew    0

Is there any way to get some documentation on this. We are interested in pursuing this, but we really cannot get anything to work. Therefore, we thought about developing our own, but there is no documentation so we could implement our own development.

 

The first issues are as follows:

 

1. When using the webrtc and pasting it in the browser, and phone will ring but there is no audio.

2. When logging into the user extension, we can initiate a call, but there is no audio.

 

 

So here are the questions:

 

1. Are you following WebRTC standards.

2. When we try creating the webrtc phone through the trunk, we cannot receive a call. When we try to use the dialpad on the screen, there is no fuctionality in the dial pad. This is when using firefox or chrome.

3. When we tried using webrtc from the user login, with a chrome browser, we can dial out but there is no audio to another extension or to pstn number when it connects. Regarding inbound calls, i search the security of the chrome brower, and i cannot find "enable/allow" this.

 

It doesnt appear that you really support WebRTC. If that is not the case, please advise by providing more information. If you are not planning to support it, could we at least get some documentation so we could develop it ourselves. We are very interested in the oppportunities regarding webrtc; however, we need some more information.

 

Thanks

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Yes we support WebRTC, but only the SDES connection (where the srtp key is exchanged in the sdp directly) and not yet the DTLS part of it.

 

That's why you have to use chrome, which supports both SDES and DTLS for now and therefore works.

 

What version of the PBX are you on?

 

Here is something about setting it up as a trunk:

 

http://vodia.com/documentation/trunk_wrtc

 

You can also use the "talk" button from our website "vodia.com" to call us for a quick test using chrome browser.

 

Thanks.

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koolandrew    0

HI,

 

I asked you some very specific questions. I dont want to bug you.

 

What is the point of hitting the talk button on your website, i have no idea what you are doing in the background.

 

We are interested in creating something similar to a softphone, using webrtc. If you cannot help us, then fine.

 

I have posted the results of our tests, its doesnt work in its present format.

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And I answered very specifically that yes it IS WebRTC but not DTLS. Even within WebRTC there are two ways, SDE and DTLS and we do SDE at this moment but not DTLS. If it's feasible we will also do DTLS but not yet. That is why only chrome will work at the moment since only chrome supports both SDE and DTLS.

 

About hitting our talk button:

 

If you are using your chrome browser WITHOUT any special plug in or download and it is able to talk to our server and you are able to hear and speak to someone, that means there is NO OTHER WAY but you were able to connect via WebRTC. That was just to show it is doing WebRTC and nothing else.

 

Signalling is not covered by WebRTC though. RFC 7118 seems to cover that and it seems interesting but that is not implemented yet.

 

I hope that was clearer.

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koolandrew    0

We would like to create a softphone based upon Webritc to work with Vodia pbx.

 

I have tested it and we cannot get it to work with chrome or otherwise, it may work on on your website, but that is really no help at this point.

 

Again, i dont want to bug you about this, but we if we could some information so we could possibly design an application to work within the constraints, that would be great.

 

The only info so far is that you support webrtc, using sde only. There is no signalling at this stage.

 

Is there anything else you can provide or is that it at this point.

 

Thanks

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koolandrew    0

HI,

 

It is almost 1.5 years since my last post, as we didnt get a response.

 

We would still like to pursue this using vodia, but i dont know to do so. Other pbx platforms are embracing this technology along with video, so i am just wondering if vodia plans on the same.

 

I realize that this may be outside of your core, and it is really only meant for talk buttons on websites, and that is fine.

 

I am just wondering if there have been any updates on Vodia's side to support Webrtc calling on the extension level.

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Vodia PBX    0

WebRTC is standard now in the PBX when you log into the web portal as a user. It is essentially a soft phone replacement. But we are not using websocket as a SIP transport layer, just JavaScript with our own little soft phone code.

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koolandrew    0

That is an interesting update.

 

I have gone to your website for an update, and there is some info.

 

1. What version of pbx is it supported from? As i cannot seem to find the release notes anymore, but i remember reading that there are some changes recently to how chrome and mozilla handle webrtc. For example, is it only from version 5.4 and on?

2. You claim webrtc is standard, for a desktop, not a mobile browser, as i dont believe there is any way to install extensions on a mobile browser.

Is there anything for a mobile browser as of yet, or is that on the horizon?

 

Therefore, my question remains.

 

Is there an api that we could utilize to try and create our own app/webpage such that we could use for webrtc.

 

Again, i dont want to keep bugging you, and if that is it, so be it.

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Although it has been supported for quite a while, of course things are constantly being added and 5.4.1 should have the most.

 

Chrome on mobile devices also does it and I think it would also be best on chrome in desktop.

 

In future we will consider using a more generic API so anyone can use it. For now you can look into the files usr_phone.js in the user portal for how we handle signalling through websockets and make the call flows for a soft phone.

 

Thanks.

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koolandrew    0

I have checked out your latest version, and it works on chrome on desktop.

 

Regarding logging in as a user from chrome on a mobile phone, or on android, i see that an image of a phone automatically appears, However, there is a media error, maybe because there is no plugin, and you probably know this already, although i dont know what "Chrome on mobile devices also does it" means from your comment above.

 

Would you happen to know if you think you will continue to use the same setup as outlined in the usr_phone.js files?

 

Thanks

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