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...other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side. I noticed that the Asterisk wiki mentions the need for res_pjsip_transport_websocket, so does that mean Asterisk requires the signaling to occur over a websocket? If I used a SIPJS fork which places the signaling over UDP (eg https://github.com/cwysong85/sipjs-udp) will it just be a regular SIP client and I shouldn't have to configure anything special in Asterisk, just regular PJSIP. -------------- next part -------------- An HTML attachment was scrubbed... URL: <...

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0....

...ers at lists.digium.com" <asterisk-users at lists.digium.com> Subject: [asterisk-users] WSS Socket Configuration Message-ID: <46B08B12850FEF43BF479468A4002A57829D17 at SZ-ORG-APP001.sabienzia.int> Content-Type: text/plain; charset="us-ascii" Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but cannot find the required configuration parameters. Does Asterisk support wss sockets? How can I configure it? Thanks, Alexej

...egister using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation ( http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. Here's my sip.conf: bindport = 5070 ;Kamailio is at port 5060, and it's...

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but cannot find the required configuration parameters. Does Asterisk support wss sockets? How can I configure it? Thanks, Alexej -------------- next part -------------- An HTML attachment was scrubbed... U...

...ble. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine, first one came directly from the client. I defined my clients according to the sip.js guide: http://sipjs.com/guides/server-configuration/asterisk/ So this was rejected: (I marked the extra lines with '//' to ease looking through the differences) v=0 o=- 9046935681162021751 2 IN IP4 91.221.66.61 s=- t=0 0 a=group:BUNDLE audio // a=msid-semantic: WMS Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF m=aud...

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The Auth user for SIP....

...ase forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions ( http://sipjs.com/guides/server-configuration/asterisk/). Calls between these work nicely without problems. Now when I call from outside, from an external Asterisk 11.5 server, I end up having problems calling from a sip client to a webrtc client. The Asterisk I have on my main testing server is the latest curr...