JsSIP

1–30 of 1155

·~· JsSIP - the JavaScript SIP library ·~·

Welcome to JsSIP mailing list.

Website: jssip.net
Mailing list address: jssip@googlegroups.com

NOTE:

Please avoid questions about Asterisk or FreeSwitch and their WebRTC implementations.

0 selected

Mike Remski

1:11 PM

RTCSession refresh, UA disconnects

Looking for some ideas. My application has long lived calls, where we hit RTCSession runSessionTimer(

unread,

RTCSession refresh, UA disconnects

Looking for some ideas. My application has long lived calls, where we hit RTCSession runSessionTimer(

1:11 PM

Andreas Backman

Mar 29

Making a one-way video call

Hello! I'm trying to place a one way video call. My CallOptions are: { rtcConstraints: {

unread,

Making a one-way video call

Hello! I'm trying to place a one way video call. My CallOptions are: { rtcConstraints: {

Mar 29

Ronald Blanco2

Mar 29

jssip latest version on react native RTCSession unmute() function it is not working

Sorry, jssip unmute is working fine, the problem seems to be with the "react-native-callkeep

unread,

jssip latest version on react native RTCSession unmute() function it is not working

Sorry, jssip unmute is working fine, the problem seems to be with the "react-native-callkeep

Mar 29

Mitul Jadav

Mar 28

(Failed / Ended) Event not fired when call ended by remote party without receive.

Hello! I am using the JSSip (Version 3.10.0) library on my project. And I'm trying to set up an

unread,

(Failed / Ended) Event not fired when call ended by remote party without receive.

Hello! I am using the JSSip (Version 3.10.0) library on my project. And I'm trying to set up an

Mar 28

Ronald Blanco, … Juan Pablo8

Mar 27

Trying to use jssip on react-native 0.70.7

Hello Ronald, So you're able to run JsSIP in RN 0.70.7? Any plans on publishing a PR to react-

unread,

Trying to use jssip on react-native 0.70.7

Hello Ronald, So you're able to run JsSIP in RN 0.70.7? Any plans on publishing a PR to react-

Mar 27

Rebecca Almeida, Ka To2

Mar 27

I can't have audio in my RTCSession.

Are you behind NAT? You might need to have STUN/TURN for audio On Mon, Mar 27, 2023 at 12:29 PM

unread,

I can't have audio in my RTCSession.

Are you behind NAT? You might need to have STUN/TURN for audio On Mon, Mar 27, 2023 at 12:29 PM

Mar 27

David Cunningham, Alex Balashov3

Mar 24

JsSIP can connect but not native Chrome

Hi Alex, Thank you for that! Have you any idea how https://tryit.jssip.net/ on Chrome manages to

unread,

JsSIP can connect but not native Chrome

Hi Alex, Thank you for that! Have you any idea how https://tryit.jssip.net/ on Chrome manages to

Mar 24

Илья Метелёв

Mar 24

Refer event is not received

During the call, we try to refer it, we get RequestSucceded and Accepted in response, but the user to

unread,

Refer event is not received

During the call, we try to refer it, we get RequestSucceded and Accepted in response, but the user to

Mar 24

Flávio Cunha

Mar 14

send call statistics on end call

Hello! How can i add a optional field with call statistics on the 200 response when i receive an

unread,

send call statistics on end call

Hello! How can i add a optional field with call statistics on the 200 response when i receive an

Mar 14

Alexandr Makovkin, … Filippo giacchè3

Mar 10

Does jsSIP support page reloading?

Reloading the page close the wesoket and the active call will be down, so i do not think is possible.

unread,

Does jsSIP support page reloading?

Reloading the page close the wesoket and the active call will be down, so i do not think is possible.

Mar 10

Christian Checcaglini

Mar 3

Call disconnect after 60 seconds if send_pai asterisk setting is set to true

Hello! i've been using this library with Asterisk 18 and we noticed a peculiar issue, if this

unread,

Call disconnect after 60 seconds if send_pai asterisk setting is set to true

Hello! i've been using this library with Asterisk 18 and we noticed a peculiar issue, if this

Mar 3

Mayfender Fender

Feb 15

Answer call one way audio

Hi All, I use Jssip connect with legacy SIP phone by using rtpengine to be the media proxy. 1. jssip

unread,

Answer call one way audio

Hi All, I use Jssip connect with legacy SIP phone by using rtpengine to be the media proxy. 1. jssip

Feb 15

Net6 Developer

Feb 8

newDTMF event not receiving remote party DTMF's

Hello! The newDTMF event is fired only for local sent DTMF's. If remote party sends DTMF during

unread,

newDTMF event not receiving remote party DTMF's

Hello! The newDTMF event is fired only for local sent DTMF's. If remote party sends DTMF during

Feb 8

Brandon Lee, Bill Kervaski6

Feb 3

_createLocalDescription answer/connect delay greater than 10 seconds

Absolute Bill! Thanks again. On Friday, February 3, 2023 at 3:20:47 PM UTC-8 bker...@heavylogic.com

unread,

_createLocalDescription answer/connect delay greater than 10 seconds

Absolute Bill! Thanks again. On Friday, February 3, 2023 at 3:20:47 PM UTC-8 bker...@heavylogic.com

Feb 3

Pikachu Pikachu

Jan 27

I dont understand how to implement DTMF

Hello! I need a help How add DTMF in this situation? Init: function initUA() { window.ua = new JsSIP.

unread,

I dont understand how to implement DTMF

Hello! I need a help How add DTMF in this situation? Init: function initUA() { window.ua = new JsSIP.

Jan 27

Tobias Wendorff

Jan 27

Chrome/Chromium allows not asking for mic

Hi there, in a project, I just wanted to listen the other side without having to give access or even

unread,

Chrome/Chromium allows not asking for mic

Hi there, in a project, I just wanted to listen the other side without having to give access or even

Jan 27

Karan Mamtora

Jan 18

react native jssip issue with release apk

when we create a release apk it doesnot register

unread,

react native jssip issue with release apk

when we create a release apk it doesnot register

Jan 18

nicolas cardone

Jan 18

transmit audio file to the other end.

hello everyone. I want to send an audio file to the other end of the call this is what i try, const

unread,

transmit audio file to the other end.

hello everyone. I want to send an audio file to the other end of the call this is what i try, const

Jan 18

ahmed mohamed2

Jan 3

jsip client with Asterisk 18 wevrtc one way audio please hepl me

sip debug <— SIP read from UDP:10.11.63.77:5068 —> INVITE sip:20...@10.12.39.223;user=phone SIP/

unread,

jsip client with Asterisk 18 wevrtc one way audio please hepl me

sip debug <— SIP read from UDP:10.11.63.77:5068 —> INVITE sip:20...@10.12.39.223;user=phone SIP/

Jan 3

Dnyaneshwer Pendurkar

Jan 2

Error: node_modules/jssip/lib/RTCSession.d.ts:1:23 - error TS2305: Module '"events"' has no exported member 'Listener'.

I am getting following issue while using jssip in angular. My angular version is 14. Can somebody

unread,

Error: node_modules/jssip/lib/RTCSession.d.ts:1:23 - error TS2305: Module '"events"' has no exported member 'Listener'.

I am getting following issue while using jssip in angular. My angular version is 14. Can somebody

Jan 2

Christian Checcaglini

12/23/22

"This tab is using your camera or microphone"

hello, i created a webphone using this application and works very well, however i got several

unread,

"This tab is using your camera or microphone"

hello, i created a webphone using this application and works very well, however i got several

12/23/22

Maxi

12/22/22

BFCP protocol implementation

Do you plan to implement?

unread,

BFCP protocol implementation

Do you plan to implement?

12/22/22

plus liao

12/13/22

'Socket' is not exported by lib/WebSocketInterface.d.ts, imported by lib/JsSIP.d.ts

When the commit id is 004d160, move the Socket and WeightedSocket of Websocket Interface.d.ts to

unread,

'Socket' is not exported by lib/WebSocketInterface.d.ts, imported by lib/JsSIP.d.ts

When the commit id is 004d160, move the Socket and WeightedSocket of Websocket Interface.d.ts to

12/13/22

Maxi, Iñaki Baz Castillo4

12/13/22

Switching the video via renegotiate don't work

What? El El mar, 13 dic 2022 a las 8:34, Maxi <prolope....@gmail.com> escribió: I am not

unread,

Switching the video via renegotiate don't work

What? El El mar, 13 dic 2022 a las 8:34, Maxi <prolope....@gmail.com> escribió: I am not

12/13/22

Hamad Mohsen

12/12/22

How can I use JsSIP with node

I'm trying to make a desktop application that supports SIP over WebSockets/Webrtc calls. However,

unread,

How can I use JsSIP with node

I'm trying to make a desktop application that supports SIP over WebSockets/Webrtc calls. However,

12/12/22

lang qiu

12/6/22

Can't get debug log any more

I can't figure out since when I can't get debug log any more. No mater I run JsSIP.debug.

unread,

Can't get debug log any more

I can't figure out since when I can't get debug log any more. No mater I run JsSIP.debug.

12/6/22

Peter James, Iñaki Baz Castillo2

11/25/22

Session

Ask in sip.js forum, not here. El El vie, 25 nov 2022 a las 11:37, Peter James <webdialer.

unread,

Session

Ask in sip.js forum, not here. El El vie, 25 nov 2022 a las 11:37, Peter James <webdialer.

11/25/22

Juan Pablo, Iñaki Baz Castillo3

11/16/22

refer event listener does not get executed after calling refer()

Yes, you're right, I'm using Asterisk. Good to know that's the problem, I'll have

unread,

refer event listener does not get executed after calling refer()

Yes, you're right, I'm using Asterisk. Good to know that's the problem, I'll have

11/16/22

Peter James

11/16/22

WebDialer hold function

Hello Guys, I am a new person joined here. I think this would be a great platform where for each

unread,

WebDialer hold function

Hello Guys, I am a new person joined here. I think this would be a great platform where for each

11/16/22

Ravhi Rizaldi

11/8/22

Connect JSSIP to FOP2 Web

Hi, How to listen/whisper using JsSIP to FOP2? Usually I can listen using click2dial (using softphone

unread,

Connect JSSIP to FOP2 Web

Hi, How to listen/whisper using JsSIP to FOP2? Usually I can listen using click2dial (using softphone

11/8/22