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Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? -- Thanks, Bhavik Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140520/82e23124/attachment.html>

...ed 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 was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/4de41094/attachment-0001.html>

...z>: > 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 both HTTPS and WSS must also come from the same >>> server...

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 ; Allow codecs in order of preference allow=ilbc

...username="770000wrtc",realm="178.18.90.230",nonce="5d5c700b",uri="sip:419 at 178.18.90.230",response="ca118222a4674b4c6dcc19dd95e00c15",algorithm=MD5 [Aug 11 15:53:47] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 [Aug 11 15:53:47] Organization: Doubango Telecom [Aug 11 15:53:47] [Aug 11 15:53:47] v=0 [Aug 11 15:53:47] o=- 5876454736929512000 2 IN IP4 127.0.0.1 [Aug 11 15:53:47] s=Doubango Telecom - chrome [Aug 11 15:53:47] t=0 0 [Aug 11 15:53:47] a=group:BUNDLE audio [Aug 11 15:53:47] a=msid-semantic: WMS kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka [Aug...

...nt-Length: 1250 Max-Forwards: 70 Authorization: Digest username="vr1a882",realm="pbx",nonce="0edd0f1f",uri="sip:887 at ipbx",response="46f10b5c84accd119fea0c65bdad3dee",algorith m=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: Doubango Telecom v=0 o=mozilla...THIS_IS_SDPARTA-38.0.1 4294967295 0 IN IP4 127.0.0.1 s=Doubango Telecom - firefox t=0 0 a=sendrecv a=fingerprint:sha-256 A4:67:26:11:1F:1E:F2:8F:75:02:FE:69:2F:FC:FA:87:7A:2C:DA:86:6D:40:43:31:B7:4C:89:0B:15:44:00:56 a=group:BUNDLE sdparta_0 a=ice-options:trickle a=msid-se...

...; needed for webrtc context=default encryption=yes dtlsenable=yes dtlsverify=no dtlsrekey=60 dtlscafile=/opt/asterisk/keys/ca.crt dtlscertfile=/opt/asterisk/keys/asterisk.pem dtlssetup=actpass insecure=invite Here is the SDP offered by Nightly: v=0 o=Mozilla-SIPUA-24.0a1 25687 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:7194cbcc a=ice-pwd:e57c14491015e529b84c5a6baf6d7b67 a=fingerprint:sha-256 48:3E:0C:59:BA:EB:6C:F9:5D:65:BF:08:54:63:C3:EA:AF:A9:60:9D:39:47:A5:41:6B:E1:A8:EB:7C:06:BE:D4 m=audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 c=IN IP4 www.xxx.yyy.zzz a=rtpmap:109 opus/48000/...

...13e6f3c96a0517b4413a6f6ce7ae;+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 2113937151 10.10.5.106 63858 typ host generation 0 a=candi...

...39;m genuinely fascinated why you are insisting on using a version of > Asterisk almost 3 years old, for which EOL support ended last year. > > Is there any particular reason you cannot or will not use the current > version as others have suggested? > > Also, I see you are using Doubango and WebRTC, but in the logs, I see > WS and WSS. > > You NEED to be using 100% WSS otherwise you've not got a hope in hell > of anything working with WEBRTC. > Check the console of the web browser you are trying to make the call > from (CTRL-SHIFT-I in Chrome on Windows, fo...

...username="770000wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419 at 178.18.90.230",response="cd2da8d1cbf0a2795b38b2048a3a3c49",algorithm=MD5 [Aug 9 22:15:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 [Aug 9 22:15:50] Organization: Doubango Telecom [Aug 9 22:15:50] [Aug 9 22:15:50] v=0 [Aug 9 22:15:50] o=- 9108976588890881000 2 IN IP4 127.0.0.1 [Aug 9 22:15:50] s=Doubango Telecom - chrome [Aug 9 22:15:50] t=0 0 [Aug 9 22:15:50] a=group:BUNDLE audio [Aug 9 22:15:50] a=msid-semantic: WMS BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps [Aug...

...sterisk",response="2ba48ca360d592ca183ba6706e6feae9",algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER > Privacy: none > P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 > User-Agent: IM-client/OMA1.0 IMSDroid/v1.2.366 (doubango r550) > P-Preferred-Identity: <sip:imsdroid> > Supported: path > > <-------------> > --- (16 headers 0 lines) --- > Sending to <my.ip.addr>:34778 (no NAT) > > <--- Transmitting (no NAT) to <my.ip.addr>:34778 ---> > SIP/2.0 100 Trying > Via...

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

...or WebRTC with Asterisk 13 and PJSIP ? > > Unfortunately, there is not much documentation about this, as far as I can > tell. > I didn't find any. > > > Regards > > [1] > https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 > [2] https://www.doubango.org/sipml5/ > > > > > Regards, > > Simon > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs:...

2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on my own private >> Asterisk server and access this demo

...brtc >> >> 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>simon.hohberg at mcs-datalabs.com>: >> >>> >>&g...

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

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