WebRTC: Ready and simple web softphone

Colleagues, can you please tell me if there is a ready-made free web softphone that can be installed on a website or customized into a program through an HTML5 window?

Of course, I installed an example from sipML5, but it’s too educational and demo. Alas, my colleagues did not like it.

Maybe there is something closer to the finished product? Really, nobody has done it yet?


I can help you with that. I am using a web phone in production.
You can contact me separately.


You can also IFRMAE it if you want:


I will be grateful. Should I send you a private message?

It’s strange that Google never found your solution for me.

Is it free, or does it have some propietary part? I saw that you have some precompiled codec libraries on your GitHub.
If I clone your GitHub section to my site, can I start using your solution completely offline?

Tell me, who did Russian localization for your solution? It was either done by a translator program without understanding the meaning of the translation, or a native speaker made a very bad joke on you.

It may be too new for google to index it well - it may need more inbound links.

It’s free, LGPL-3.0 License. The included libraries (hosted via CDN) are also free, like jquery, sipjs, etc are also free (LGPL - refer to each).
The setup instructions are based on Asterisk with a very very very basic dial plan - so you can do as you like on top of that - so long as you understand the concepts of dialplan etc you should be fine.

No, the pre-complied codec was just opus - its more tricky with the ARM cpu, as Asterisk don’t supply a compiled version during install (its x86 based only).
(For simplicity the video’s etc are based on Raspberry Pi, but if you are familiar with other systems, and can get Asterisk going, just go ahead.)

There is a library pack that includes all the files that the system downloads during use, so it can operate “offline”… but remember a PeerConnection MUST have a STUN server… don’t ask why, I find it annoying too… but anyway, it will try to find your “live” ip so that RTP can be established - so in an offline solution, this STUN lookup will fail after 500ms.

Oh!? lol… no some guy said he would help… I have no idea what he wrote. I guess this is the entire point about doing an open source githup project… people can jump in and help.

As I understand it, they are posted on http://dtd6jl0d42sve.cloudfront.net?

Is there a way to clone them to my site without interfering with updating the main code?


And I did not find the setting at all that sets the path to the STUN server. How can it be explicitly set?


It’s on the SessionDescriptionHandler (https://sipjs.com/api/0.11.0/sessionDescriptionHandler/)

But let me make it a setting, then you can use your own over your LAN.

(btw, if you want, you could also create a static DNS entry for stun.l.google.com and just point it.

There is a link the the downloadable library pack on the GitHub home page.

One paragraph is not translated correctly and is an obscene word. If you say it in public, the police will detain you. And the judge will arrest you for fifteen days. (For all the Russian dislike for the authorities, public opinion will be on their side.)
Many other translations are reminiscent of the conversations of Scary Russian Villains from Hollywood movies. It’s very funny, but it’s not clear what it means.

I have installed your solution on my site and am editing the localization file little by little. If you will accept my corrections - I’m will be flattered …

In addition, I had several comments and suggestions. Are you interested in this?


Nice :slightly_smiling_face:!

Lol My goodness… I should try check that some how!

Please feel free to update. Fork it, and apply the changes to the language file and send a pull request. I’m currently busy with the main phone.js file, so any changes there will probably conflict.

Also fell free to make any suggestions. If you don’t mind to use the GitHub Issues page for these:

I created several themes there.


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