JsSIP

At some point you have to get SIP traces or logs in your SIP proxy/server and see if the ACK reaches

yes, asterisk 16.8-cert8 we added contact_uri option to UA config contact_uri: `sip:${this.sipValue}@

Hi My application was working normally, suddenly I couldn't get the remote video in both ends, in

Hi, there is currently no such feature. Out of curiosity, which Contact header parameters do you want

Hello everyone. I'd appreciate a lot your help with this issue. I am using Kazoo platform (https:

Double NAT? From: <jssip@googlegroups.com> on behalf of AD Rehman <adrehman@gmail.com>

I've got the exact same Problem -.- still searching for an solution... In some forums describing

You can ask in the sipjs forum. On Thu, 8 Jul 2021 at 09:08, Sureshstephen Sugumar <

I'm afraid doing audio recording is not JsSIP business. On Fri, 2 Jul 2021 at 12:04, AD Rehman

Hello, if you are in a call and the page is refreshed, its possible return to the initial call to

SIP sessions are established correctly as per your comments, which is JsSIP's duty. You are

Still down for me On Sunday, June 13, 2021 at 12:37:45 PM UTC-4 tiya...@gmail.com wrote: It looks

Hi mukesh, Thanks for the reply. Can you please guide me how I can implement conference call ? We

Hello Guys, I'm facing the same issue of making conference call. I tried above code but I think

You can tell your colleagues that JsSIP is configured correctly and Asterisk has wrong configuration

I'm afraid this is not a JsSIP related issue or question. El El jue, 17 jun 2021 a las 12:03,

Found this: JsSIP.Socket interface: Event Handlers: ondata(data) -> Must be called when there is a

Well, I managed to figure something out. A closer reading of node js documentation led me to "

Hello. I have a problem. I don't understand the logic of incoming calls. What methods from the

Hi, You can provide JsSIP a specific Socket implementation conforming the Socket[*] interface. Within

This is not a JsSIP question but a generic WebRTC question so you may get better answers in a webrtc

OK. It's workaround, not solution. I think in the forum described how limit ICE gathering time -

Alex, I made a test between baresip client on Android and JsSIP Tryit client on Firefox. baresip

You have the solution in that thread and you should check the JsSIP API documentation. El El dom, 2

JsSIP is a client side SIP UR library. And this is a mailing list about JsSIP rather than about SIP

I am trying to implement the same. Anyone with an idea on how this can be implemented ? On Wed, Apr

is there a way to stream audio from nodejs? -- Thanks. Kind Regards, Lee M. Lwando +260950482560 CTO

Sure. Then ask in PJSIP mailing list. El El mié, 21 abr 2021 a las 5:21, Алексей Карсаков <

Just learn how the debug module works and the API it exposes for changing the output. El El vie, 16

Hello! I'm using JSIP 3.7 and i need to capture remote audio in the answer method. I saw that is