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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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Vianney Lejeune
2
May 25
peerconnection event not firing on outbound call?
Guys, I think the problem is due to the audio stream not being assigned to the HTML audio object.
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peerconnection event not firing on outbound call?
Guys, I think the problem is due to the audio stream not being assigned to the HTML audio object.
May 25
Cheng Lin Yu
2
May 22
"socket: WebSocketInterface, error: true, code: 1006, reason: ''
Has anyone solved this problem? It's really urgent 在2023年5月23日星期二 UTC+8 11:45:32<Cheng Lin Yu
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"socket: WebSocketInterface, error: true, code: 1006, reason: ''
Has anyone solved this problem? It's really urgent 在2023年5月23日星期二 UTC+8 11:45:32<Cheng Lin Yu
May 22
Vianney Lejeune
, …
Bill Kervaski
4
May 18
Demo looks down
You can just roll your own, lots of open source options. From: <js...@googlegroups.com> on
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Demo looks down
You can just roll your own, lots of open source options. From: <js...@googlegroups.com> on
May 18
Cheng Lin Yu
2
May 11
socket: WebSocketInterface, error: true, code: 1006, reason: ''当频繁刷新去连接信息命令服务时,收到一个新的来电,去连接听这个通话,同时此时信息命令服务事务正断开,引导致无法经常接听,如何规避这个问题呢?
当频繁刷新浏览器去重新连接通信命令服务器时,接收到电话通话,这个时候经常会遇到服务中断websocket断开连接的情况,导致无法正常去answer接听通话,知道是什么原因导致的这个结果吗,或者如何规避
unread,
socket: WebSocketInterface, error: true, code: 1006, reason: ''当频繁刷新去连接信息命令服务时,收到一个新的来电,去连接听这个通话,同时此时信息命令服务事务正断开,引导致无法经常接听,如何规避这个问题呢?
当频繁刷新浏览器去重新连接通信命令服务器时,接收到电话通话,这个时候经常会遇到服务中断websocket断开连接的情况,导致无法正常去answer接听通话,知道是什么原因导致的这个结果吗,或者如何规避
May 11
Mike Remski
,
José Luis Millán
2
May 10
Malformed Authorization header?
The JWT is sent in the Authorization SIP header of the messages. El mié, 5 abr 2023 a las 17:32, Mike
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Malformed Authorization header?
The JWT is sent in the Authorization SIP header of the messages. El mié, 5 abr 2023 a las 17:32, Mike
May 10
Gabriel Barreto
Apr 20
Does Jssip work with react native anyway?
Does Jssip work with react native anyway? who could help with that? I read some comments here in the
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Does Jssip work with react native anyway?
Does Jssip work with react native anyway? who could help with that? I read some comments here in the
Apr 20
J Arsh
Apr 20
the RTCSession 'failed' event's callback func params has incorrect type using TypeScript
if local reject the calling,`faild` event's callback function return { message:null } but in the
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the RTCSession 'failed' event's callback func params has incorrect type using TypeScript
if local reject the calling,`faild` event's callback function return { message:null } but in the
Apr 20
Ivan Hutomo
, …
mah kho
4
Apr 17
[TypeError: null is not an object (evaluating 'constraints.optional')] when call sipUa.call
u need to handle get permission first if u want i can help you i had this issue before On Thu, Apr 13
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[TypeError: null is not an object (evaluating 'constraints.optional')] when call sipUa.call
u need to handle get permission first if u want i can help you i had this issue before On Thu, Apr 13
Apr 17
chetan jha
Apr 13
Optional Parameter
Hello All, I just wanted to check if we can declare an optional parameter to get information from
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Optional Parameter
Hello All, I just wanted to check if we can declare an optional parameter to get information from
Apr 13
Mike Remski
Apr 3
JsSIP::UA::set authorization_jwt
Morning folks. A quick question about the setter for JsSIP UA for authorization_jwt. If I call that
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JsSIP::UA::set authorization_jwt
Morning folks. A quick question about the setter for JsSIP UA for authorization_jwt. If I call that
Apr 3
Ravhi Rizaldi
Apr 2
How to disable Comfort Noise?
Hello, My asterisk got a log like this below : "NOTICE [22152] [C-0000000c]: res_rtp_asterisk.c:
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How to disable Comfort Noise?
Hello, My asterisk got a log like this below : "NOTICE [22152] [C-0000000c]: res_rtp_asterisk.c:
Apr 2
Mike Remski
Mar 30
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(
Mar 30
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
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 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