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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
,
José Luis Millán
4
May 17
RTP Timeout
On Tue, 17 May 2022 13:12:21 +0200 'José Luis Millán' via JsSIP <jssip@googlegroups.com
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RTP Timeout
On Tue, 17 May 2022 13:12:21 +0200 'José Luis Millán' via JsSIP <jssip@googlegroups.com
May 17
Nouman Ansari
,
José Luis Millán
2
May 17
JSSIP Fail over behaviour
Reconnection will happen in random interval times between the given config values: https://jssip.net/
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JSSIP Fail over behaviour
Reconnection will happen in random interval times between the given config values: https://jssip.net/
May 17
Orion project
2
May 16
From README example script: REGISTER ok, CALL nothing happens
Ok it seems that my pbx needs an ACL to allow media to start. I modified the example code to show to
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From README example script: REGISTER ok, CALL nothing happens
Ok it seems that my pbx needs an ACL to allow media to start. I modified the example code to show to
May 16
Mike Remski
May 9
ICE, multiple interfaces, RTP timeouts
Looking for a few ideas of where to start looking. Node.js application, not running in a browser,
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ICE, multiple interfaces, RTP timeouts
Looking for a few ideas of where to start looking. Node.js application, not running in a browser,
May 9
Filippo giacchè
3
May 6
Call forward
Just for completness... Finaly I have found a solution that seem work. I used the terminate method
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Call forward
Just for completness... Finaly I have found a solution that seem work. I used the terminate method
May 6
JRed
May 3
Interconnection with SIP Server
Hi, Our initial idea to connect JsSIP WebRTC client to a SIP Server / IMS was using Asterisk. However
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Interconnection with SIP Server
Hi, Our initial idea to connect JsSIP WebRTC client to a SIP Server / IMS was using Asterisk. However
May 3
Александр Чёрный
Apr 29
Emit sip notify event from one user agent to another user agent
Hi. I'm trying to create a functionality when one softphone register, another one with the same
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Emit sip notify event from one user agent to another user agent
Hi. I'm trying to create a functionality when one softphone register, another one with the same
Apr 29
Murugan Pandian
Apr 26
ICE Gathering Fails
ICE Gathering fail on chrome 100.0.4896.127 (MAc) Issue: When we try to make calls we are getting an
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ICE Gathering Fails
ICE Gathering fail on chrome 100.0.4896.127 (MAc) Issue: When we try to make calls we are getting an
Apr 26
Anton Alberdi
Apr 25
Setting Caller ID for a call
I have two numbers in my trunk line. Is there any method or header to set up the caller id of my
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Setting Caller ID for a call
I have two numbers in my trunk line. Is there any method or header to set up the caller id of my
Apr 25
Paul Koenig
Apr 24
Refer-To Replaces From/To Tag Selection
Hello, I am looking for some guidance on attended transfer Refer-To, and whether the below might be a
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Refer-To Replaces From/To Tag Selection
Hello, I am looking for some guidance on attended transfer Refer-To, and whether the below might be a
Apr 24
Mir Garey
,
Alex Balashov
4
Apr 20
answer() method doesn't work on Windows 11
Yeah… There's WebRTC, and then there's the outermost “user interface” to WebRTC, which sadly
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answer() method doesn't work on Windows 11
Yeah… There's WebRTC, and then there's the outermost “user interface” to WebRTC, which sadly
Apr 20
Daniel Besoli
Apr 13
Asterisk KEEPALIVE
Hello guys I have a question about a setup done with asterisk. I receive every 15 seconds an empty
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Asterisk KEEPALIVE
Hello guys I have a question about a setup done with asterisk. I receive every 15 seconds an empty
Apr 13
Pavel Balashov
Apr 8
Trickle ICE support (RFC8840)
Hello, Looking through codebase and examples I've got the impression that usage of Trickle ICE in
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Trickle ICE support (RFC8840)
Hello, Looking through codebase and examples I've got the impression that usage of Trickle ICE in
Apr 8
Raman Ayyar
Apr 7
ReInvite is not send after ICE negotiation for DTLS connectivity
Application is running in node.js. JSSIP sends INVITE with local candidates, and receives 200 OK from
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ReInvite is not send after ICE negotiation for DTLS connectivity
Application is running in node.js. JSSIP sends INVITE with local candidates, and receives 200 OK from
Apr 7
Alberto Santamaria
Apr 7
question: is it possible to establish a sip p2p connection with jssip but no sip server?
Hello, First of all, thank you so much to all contributors, I´ve been using the jssip library for a
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question: is it possible to establish a sip p2p connection with jssip but no sip server?
Hello, First of all, thank you so much to all contributors, I´ve been using the jssip library for a
Apr 7
Mike Remski
, …
Raman Ayyar
3
Apr 5
488 response after receving 200 OK
Thanks, SDP in 200 OK had extra JsSIP:Transport a=rtcp-fb:111 transport-cc After removing that, the
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488 response after receving 200 OK
Thanks, SDP in 200 OK had extra JsSIP:Transport a=rtcp-fb:111 transport-cc After removing that, the
Apr 5
ryan embgrets
,
Jehanzaib Younis
2
Apr 5
Failed to set remote answer sdp: The order of m-lines in answer doesn't match order in offer. Rejecting answer.
Hi Ryan, Did you find the solution yet? I am having the same error. Can you please describe the
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Failed to set remote answer sdp: The order of m-lines in answer doesn't match order in offer. Rejecting answer.
Hi Ryan, Did you find the solution yet? I am having the same error. Can you please describe the
Apr 5
lang qiu
Apr 2
Re-Register during a phone call but with different call-id
Hi, I am experiencing a problem that during a phone call, JsSip re-register with a new call-id, so my
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Re-Register during a phone call but with different call-id
Hi, I am experiencing a problem that during a phone call, JsSip re-register with a new call-id, so my
Apr 2
Raj
2
Mar 24
UA Request-URI does not point to us
Did some more digging into the source code of JsSIP. The condition that triggers the error message is
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UA Request-URI does not point to us
Did some more digging into the source code of JsSIP. The condition that triggers the error message is
Mar 24
Alpt
,
Aleksandr
2
Mar 24
IOS15 websocket error
Did you manage to solve this problem? воскресенье, 19 сентября 2021 г. в 03:48:11 UTC+3, Alpt: Hi All
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IOS15 websocket error
Did you manage to solve this problem? воскресенье, 19 сентября 2021 г. в 03:48:11 UTC+3, Alpt: Hi All
Mar 24
Terry Don Bartels
Mar 19
Test websocket connection
Is it possible to test the websocket connection in an app like postman before implementing in JsSip I
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Test websocket connection
Is it possible to test the websocket connection in an app like postman before implementing in JsSip I
Mar 19
JRed
,
Alex Balashov
3
Mar 17
NAT traversal
That was quick ! Thank you ! Will try. El jueves, 17 de marzo de 2022 a las 13:40:37 UTC+1, abal...@
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NAT traversal
That was quick ! Thank you ! Will try. El jueves, 17 de marzo de 2022 a las 13:40:37 UTC+1, abal...@
Mar 17
rhonda
,
Igor K
2
Mar 9
Play dtmf sounds to user
You can play DTMF sound using AudioContext https://developer.mozilla.org/en-US/docs/Web/API/
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Play dtmf sounds to user
You can play DTMF sound using AudioContext https://developer.mozilla.org/en-US/docs/Web/API/
Mar 9
Алексей Макуш
,
Alex Balashov
2
Mar 8
Accept a call/Miss a call
Привет Алексей, Both inbound and outbound calls generate an event, on the JsSIP.UA level, called
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Accept a call/Miss a call
Привет Алексей, Both inbound and outbound calls generate an event, on the JsSIP.UA level, called
Mar 8
Metehan Metin
Mar 2
on reinvite setLocalDescription happening before addtrack
Hello I am trying to implement renegotiation with reinvite. 1 - starting an audio call 2- adding
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on reinvite setLocalDescription happening before addtrack
Hello I am trying to implement renegotiation with reinvite. 1 - starting an audio call 2- adding
Mar 2
Sido
Feb 27
backoff algorithm
I am confused about the usage of the following parameters. connection_recovery_min_interval: 5,
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backoff algorithm
I am confused about the usage of the following parameters. connection_recovery_min_interval: 5,
Feb 27
Nathan Stratton
,
Iñaki Baz Castillo
2
Feb 13
error parsing header "Contact" when using IPv6
> Contact: <sip:31FFCF4D-62095C3A0004642B-0D8D8700@2602:fc11::31:443;transport=ws> I'm
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error parsing header "Contact" when using IPv6
> Contact: <sip:31FFCF4D-62095C3A0004642B-0D8D8700@2602:fc11::31:443;transport=ws> I'm
Feb 13
Jeff Whelpley
Jan 25
Listen-only session not working in Safari or Firefox
I am using jssip to make a call where the user is only listening to the conference audio from a
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Listen-only session not working in Safari or Firefox
I am using jssip to make a call where the user is only listening to the conference audio from a
Jan 25
s fan
Jan 25
jssip authentication did not initiate the second registration #761
As shown in the figure above, jssip is always timeout in the first registration process, After the
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jssip authentication did not initiate the second registration #761
As shown in the figure above, jssip is always timeout in the first registration process, After the
Jan 25
Navdeep Garg
Jan 25
jssip@googlegroups.com
Hii, we are making project in react-native v0.63.3 using this this library. when we making call
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jssip@googlegroups.com
Hii, we are making project in react-native v0.63.3 using this this library. when we making call
Jan 25