sipml5 | The world 's first HTML5 SIP client | SDK library

by DoubangoTelecom JavaScript Version: 1.4.217 License: BSD-3-Clause

X-Ray Key Features Code Snippets Community Discussions(7)Vulnerabilities Install Support

kandi X-RAY | sipml5 Summary

sipml5 is a JavaScript library typically used in Utilities, SDK applications. sipml5 has no bugs, it has no vulnerabilities, it has a Permissive License and it has medium support. You can download it from GitHub.

Project home page: sipml5.org Download JS API: SIPml-api.js.

Support

Quality

Security

License

Reuse

Support

sipml5 has a medium active ecosystem.

It has 877 star(s) with 437 fork(s). There are 94 watchers for this library.

It had no major release in the last 6 months.

There are 206 open issues and 146 have been closed. On average issues are closed in 87 days. There are 6 open pull requests and 0 closed requests.

It has a neutral sentiment in the developer community.

The latest version of sipml5 is 1.4.217

Quality

sipml5 has 0 bugs and 0 code smells.

Security

sipml5 has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

sipml5 code analysis shows 0 unresolved vulnerabilities.

There are 0 security hotspots that need review.

License

sipml5 is licensed under the BSD-3-Clause License. This license is Permissive.

Permissive licenses have the least restrictions, and you can use them in most projects.

Reuse

sipml5 releases are not available. You will need to build from source code and install.

sipml5 saves you 8304 person hours of effort in developing the same functionality from scratch.

It has 17061 lines of code, 2 functions and 228 files.

It has low code complexity. Code complexity directly impacts maintainability of the code.

Top functions reviewed by kandi - BETA

kandi has reviewed sipml5 and discovered the below as its top functions. This is intended to give you an instant insight into sipml5 implemented functionality, and help decide if they suit your requirements.

Parse TIP .
Executes a TSIP message from a TIP .
Constructs a new Session Dialog
publish output for a JSDoc set
invoked when the invite request has received from the media server
Run animation
Handle the responses .
Main function .
Handler for changes to X2X .
Parses an ICAP instance .

Get all kandi verified functions for this library.

sipml5 Key Features

No Key Features are available at this moment for sipml5.

sipml5 Examples and Code Snippets

No Code Snippets are available at this moment for sipml5.

Community Discussions

Trending Discussions on sipml5

Asterisk: WebRTC no audio

asterisk Call ID in sipml5

How to configure REFER call in SIPML5 WebRTC?

JSSIP and React audio issue

How to add external javascript library in a gateway project

WebRTC websocket SSL certs use to work but not any more? ERR_INSECURE_RESPONSE

Sipml5 errors when running inside of Phoenix frameworks

QUESTION

Asterisk: WebRTC no audio

Asked 2020-Jan-07 at 13:59

I'm having what is probably a simple configuration issue. Calls between two SIP clients (zoiper) are successful.

When I start a call between a WebRTC client (sipml5) and a SIP client (Zoiper) is the connection active, but there is no audio in both directions available.

So, I have latest Asterisk 16, latest Chrome (with Firefox & Chrome Beta the same problem), sipml5 and a local network - no nat or firewall.

Any ideas on what we may be doing wrong? Thank you in advance for any help!

/etc/asterisk/sip.conf

...

ANSWER

Answered 2020-Jan-07 at 13:59

SOLVED: The problem was an AD-Block Plugin.

Source https://stackoverflow.com/questions/59361965

QUESTION

asterisk Call ID in sipml5

Asked 2018-Aug-23 at 15:03

How do I get asterisk call Id (uniqueid in cdr table) (for instance, 1487150355.465) in sipml5 client. As far as I looked, I see only https://www.doubango.org/sipml5/docgen/symbols/SIPml.Session.html#getId which has (afaic) no relation to asterisk id.

I know I can set additional headers in asterisk and set there call id, but it cannot be done for some organizational reasons. Thanks.

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ANSWER

Answered 2017-Mar-10 at 21:12

Asterisk unique channel id(not call id) assigned to channel. By default asterisk not send that via sip or other protocol.

So no, you can't get it without do something on asterisk.

Source https://stackoverflow.com/questions/42726760

QUESTION

How to configure REFER call in SIPML5 WebRTC?

Asked 2018-Aug-02 at 06:24

I am trying to make a web client for my SIP call request. I have done invite call successfully from browser. But, I am not getting how to refer to the 3rd party call through Javascript. I am using WebRTC and SIPML5.

On trying session.transfer(),after successful Refer, the link to the referrer is broken... but, I am able to hear audio from the referee to referrer. 'I am not getting Notify. Just refer is ACK sip message and then 1 sided call.'

For example: A refers to B. After successful refer, I am not able to send/recv SIP packets from A. But, I am able to do the same from B. So, when B talks, A can hear. But, vice-versa is not working.

The SIP refer is working perfectly via SIPP. And 2-way packets are available.

P.S.: I DONT WANT CALL TRANSFER.

If SIPML5 does not support this, can I do the same via Java. Where I send a refer request to SIP servelet which handles it via doRefer()?

These are the links I referred:

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ANSWER

Answered 2018-Aug-02 at 06:24

I found it on my own.

SIPML5 does not support call refer but call transfer. So, it uses REFER to transfer call but then, the callee gets disconnected.

So, finding a workaround, I sent the REFER message details in session.send() as a string. I had to parse the string to fetch the data which ofcourse is not the SIP message but a simple string.

and on handling refer, I am able to receive packets from callee and caller.

The problem persists though,

The problem is that the invite call's session is of type 'call' whereas the message session was of type 'message'.

So, I added CALL-ID of call session into Message session and parsed it. Thus successfully getting the packets on same channel.

Source https://stackoverflow.com/questions/50924893

QUESTION

JSSIP and React audio issue

Asked 2018-Jun-28 at 17:54

So I am using jssip 3.2.10 to make calls on a React project.

The server is setup on Asterisk and CentOS.

I can make calls where the call receiver hears me well, but I can't hear their audio, nor the waiting (traditional) beep noises it should make until the call is picked up.

It does work with some sipml5/asterisk udp online tests so I feel it's on my clients side issue. I tested it on Chrome and Firefox (both latest, with the same results).

My setup

I have a helper to connect called sip.js:

...

ANSWER

Answered 2018-Jun-28 at 17:54

Thanks to the answer here: How to handle audio stream in JsSIP?

I found the solution, I needed to add to the file rendering the call:

Source https://stackoverflow.com/questions/51073157

QUESTION

How to add external javascript library in a gateway project

Asked 2017-Aug-28 at 22:46

I'm trying to add Simpl5 javascript library to my gateway unsuccessfully. I've put SIPml-api.js and SIPml.js in webapp/content/scripts folder.

In .angular-cli.json I've update scripts array like this :

...

ANSWER

Answered 2017-Aug-28 at 22:46

JHipster does not fully supports angular cli as @GaëlMarziou already mentioned in comments.

The original idea of supporting the CLI was only for code generation.

So any .angular-cli.json modificatons won't have any effect as JHipster is using its own build/serve chain (have a look in to package.json custom scripts, they are not based on ng serve/build, etc ) check this issue discussion for more details.

You need to reference your custom css and js in vendor.ts then when you run npm run serve you will see the result.

Source https://stackoverflow.com/questions/45898654

QUESTION

WebRTC websocket SSL certs use to work but not any more? ERR_INSECURE_RESPONSE

Asked 2017-Jun-02 at 17:55

I am using sipml5 to register sip endpoints and using sipml5 as my client and FreeSwitch as my SIP server. I use to be able to register my sip endpoint and establish a connection, but recently I been getting this error in my chrome and Firefox dev console "ERR_INSECURE_RESPONSE Error in connection establishment". I looked at probably causes and was not able to solve my problem. When I change my proxy to using ws instead of wss it works. I am using SHA256 algorithm in my SSL cert. I added my domain name to SubjectAltName as well as my CN in my SSL cert as well. I am using openssl to create these certs.

What else could be the cause of this error.

...

ANSWER

Answered 2017-Jun-02 at 17:55

Wherever the websocket server lives (wss://my_ip:7443) needs to have a proper certificate. For now, you can manually accept the certificate by typing https://my_ip:7443 into your browser.

Toye

Source https://stackoverflow.com/questions/44293870

QUESTION

Sipml5 errors when running inside of Phoenix frameworks

Asked 2017-Feb-25 at 22:36

When I am running sipml5 in Phoenix framework. I got errors in below. The problem is that I have no error when I use sipml5 without frameworks. I tested this in Flask and Django too. But I got the same errors again.

...

ANSWER

Answered 2017-Jan-12 at 14:59

I solved the problem. Unfortunately, phoenix (in my case) couldn't support separate folders which is required to run of api. So for this reason you should need to rewrite your code and use apartly.

Source https://stackoverflow.com/questions/41522287

Community Discussions, Code Snippets contain sources that include Stack Exchange Network

Vulnerabilities

No vulnerabilities reported

Install sipml5

You can download it from GitHub.

Support

For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .

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