Stack Overflow
Stack Overflow

Questions tagged [sipjs]

A simple yet powerful JavaScript library that takes care of WebRTC and SIP signaling for you.

Features

  • Register SIP User Agents using the SIP over WebSocket transport
  • Create Audio and Video sessions
  • Send Instant Messages and view Presence
  • Share your screen or desktop
  • Utilize advanced call features such as early media, call hold and resume, and transfers
  • Send DTMF with SIP INFO
  • 100% open source, 100% JavaScript
  • All major browsers supported

Homepage: https://sipjs.com/

33 questions
0
votes
0 answers

Is there any way to use sipjs or other method to get rtcp packets？

I use sip.js to mkae a webrtc test. And I wang to get some information from rtcp. I tried to use getStatus() API,but I want the rtcp packet form the other side.Because getStatus() can not calculate the out-bound rtp packt lost and jitter. Is there…
cqupt_zz
  • 1
  • 1
0
votes
1 answer

How can I use sip.js connect freeswitch with wss

I am trying to register with the server using sipjs.It can works with ws,but when i use wss,some errors occured.I have used a self-signed certificate wss.pem to instead the old one freeswitch error tport.c:2777 tport_wakeup_pri()…
cqupt_zz
  • 1
  • 1
0
votes
0 answers

How audio is sent in sipjs?

I am using sip.js to establish an audio call, but I can't understand how the audio is sent and works. I tried to read the documentation but I didn't found where it is mentioned regarding it.
Naveen Kumar
  • 98
  • 7
0
votes
0 answers

getting an error while trying to use an inviter.invite() function in sip.js

When I try to use the inviter.invite() function from sip.js, I get an error related to mediastream not defined and session-description-handler. userAgent.start().then(() => { const target = UserAgent.makeURI("sip:1000@ip_address:5060"); const…
Jeet Detroja
  • 1
  • 2
0
votes
0 answers

How to recover SIP js Invitation Object or Session Object in React Js on page refresh

I am implementing Audio/Video call with SIP js and Astrisk server in React JS.I was successful on creating the WebRTC Audio/Video calling. But I am facing an issue with storing the Invitation or Session Object for SIP js. Because Circular JSON data…
Ishaque Javed
  • 406
  • 4
  • 14
0
votes
1 answer

Does chrome support audio codec G722,G723,G729?

Recently, when using sip.js to write a webrtc project, we cannot use the G722,G723 or G729 to make calls.Does Chrome support these voice codes？ I think G722 is workable. How about the others?
cqupt_zz
  • 1
  • 1
0
votes
1 answer

freeswitch contact header (null) user

i have a freeswitch server and am using external profile to register extension 1000 from SIP.js the connection goes well and i am able to see auth etc. passing through, however the response of registration has incorrect Contact header. it's missing…
shwetank
  • 115
  • 3
  • 7
0
votes
2 answers

ICE restart with SIP.js

What is the proper procedure to do ICE restart when using SIP.js? (v0.20.0) This is what I'm trying: oniceconnectionstatechange: (event) => { const newState = sdh.peerConnection.iceConnectionState; if (newState == 'failed') { …
Janis
  • 53
  • 6
0
votes
0 answers

SIP js UserAgent API, I don't kown how to establish relationship with dom element, version 0.15x

I don't kown how to establish relationship with dom element, version 0.15x, in 0.06x version it's options.media.render ,but what's it in 0.15x
…
fbfatboy
  • 371
  • 1
  • 6
0
votes
1 answer

How to send (Screen sharing stream) via SIPJS to the other caller

I'm using SIPJS to make calls between 2 callers using web browser. Now i want to add (Screen sharing) feature , so far i managed to open chrome screen sharing window and i get the stream and played it in video element. But what i really need is to…
Ahmad Okaily
  • 84
  • 10
0
votes
1 answer

How to get video streams from FusionPBX video conference room

I'm using SIP.js to connect to FusionPBX video conference room, but when callers join, all callers get a single caller video stream. How can I get all streams in the room to handle them and view them to each others so everyone can see everyone…
Ahmad Okaily
  • 84
  • 10
0
votes
1 answer

Where should be the location for coturn or ice setting for sipjs 0.11.0?

I am moving from sipjs 0.7x to sipjs 0.11 After reading the Git issue https://github.com/onsip/SIP.js/pull/426#issuecomment-312065734 and https://sipjs.com/api/0.8.0/sessionDescriptionHandler/ I have found that the ice options (coturn, turn, stun)…
Leon
  • 328
  • 2
  • 5
  • 13
0
votes
1 answer

Migration sipjs to jssip

I changed lib sipjs to jssip. I have problem on session transfer. in sipjs this look like this session_from.refer(session_to); How i can do this on jssip?
UJin
  • 5
  • 6
0
votes
1 answer

SipJS and Freeswtch : Not able to receive phone call

Trying to call using Freeswitch and sipJS based SipPhone I am using linphone at one end and sipjs at another , lin phone is able to call browser bases sipJs phone as its ringing but I am not able to receive call Here Are My Logs, Unauthorized UA…
Satvik Purohit
  • 41
  • 5
0
votes
2 answers

Make an attended transfer with SIP.js

Good morning, I need to make attended transfers with SIP.js. Anyone succeded in this task? I can only make blind transfers right now, i found an article that reports that in version 0.7.x there is support for attended transfer trough replace…
CaTourist
  • 711
  • 3
  • 9
  • 21
1
2
3