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JsSIP
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·~·
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.
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Mike Remski
1:11 PM
RTCSession refresh, UA disconnects
Looking for some ideas. My application has long lived calls, where we hit RTCSession runSessionTimer(
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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: {
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Making a one-way video call
Hello! I'm trying to place a one way video call. My CallOptions are: { rtcConstraints: {
Mar 29
Ronald Blanco
2
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
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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
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(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 Pablo
8
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-
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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 To
2
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
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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 Balashov
3
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
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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
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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
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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 giacchè
3
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.
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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
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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
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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
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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 Kervaski
6
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
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_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.
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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
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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
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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
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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 mohamed
2
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/
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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
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"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?
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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
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'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 Castillo
4
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
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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,
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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.
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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 Castillo
2
11/25/22
Session
Ask in sip.js forum, not here. El El vie, 25 nov 2022 a las 11:37, Peter James <webdialer.
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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 Castillo
3
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
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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
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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
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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