search for: sipml5

Hello All, I am using Asterisk 12 and sipml5 as front-end and when i call from one to another the call will ring on other end but when i allow the camera access call will terminated automatically. I have attached the logs of Asterisk, if some one will get something useful Please reply on the same. Thanks and Regards, Anant == Using...

Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support . I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and als...

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Al...

users are registering over ws:// but while dialing A -> B , using either jssip/sipml5 I receive an error on B side saying , ice related information is missing , and in INVITE sdp mentioned fields are really missing. Exact error in console is SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd lots of users are using webrtc with jssip/sipml5 + asterisk su...

Hi everyone, I'm currently trying to play a little with WebRTC using sipml5 client and Asterisk trunk (370821) It seems the the WebRTC implementation for Asterisk 11 is already available in the trunk? Am I right? http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html I'm having trouble to even register to my Asterisk server using sipml5 client. The only...

...ocket': An insecure WebSocket connection may not be initiated from a page loaded over HTTPS.* If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws, this error message becomes with *Disconnected: Failed to connet to the server* My questions are: 1. Is wss now required by sipml5 live demo (implying wiki page is not up-to-date) ? 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ? Regards [1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 [2] https://www.doubango.org/sipml5/ -------------- next part -------------- An HTML attachment...

Hi all ! I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in order to test WebRTC setup on my Asterisk PBX. I am using latest SVN version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677) If I make calls from softphones (Zoiper, X-Lite), which do not support DTLS at all, I can hear the Echo Test sound. BUT when I call from br...

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk...

...016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo ( https://www.doubango.org/sipml5/call.htm?svn=241) ? > Dne 18.2.2016 v 15:36 Olivier napsal(a): > > > > 2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > >> >> Is it implied here that b...

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

...yone 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 Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and Asterisk rejects it with "We are requesting SRTP for audio, but they re...

...er=yes;click2call=no;transport=ws>;impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language="en,fr,it" Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc CSeq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5 v=0 o=- 365893986064703740 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 m=audio 37874 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 85.0.XXX.XXX a=rtcp:37874 IN IP4 85.0.XXX.XXX a=candidate:296123718 1 udp 2...

Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the asterisk server my laptop being 192.168.241.10 and asterisk being 192.168...

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance '0x7f3ccc020718'" I am struck here. Please throw some light to go further. Thanks in advance. Best rega...

Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After that, I'm only seeing "Access Denied" web page. "You do not ha...

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101 --> Asterisk sends "SIP/2.0 488 Not acceptable here" Chrome: I've tried both si...

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio...

tengo la siguiente pagina pero no se como seguir despues del punto 22 http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html gracias! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141112/59751a87/attachment.html>

...9;ll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent things to work. [1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 > > i want try jssip > https://github.com/versatica/JsSIP > it looks like a lot "livelier" than sipml5 > > any experience with jssip? > > > Dne 18.2.2016 v 16:01 Olivier napsal(a): > > > > 2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at f...