similar to: DTLS setting impacts encryption setting

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

I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017) and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome. I have seen other people post a similar issue, but I have not seen a solution. If someone with good knowledge of this issue were to respond with "this is a known

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 browser (I've tried latest Mozilla Firefox

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

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly, will ast negotiate srtp or dtls even ast and the proxy speak sip in the clear over the lo interface? Avoiding encryption over lo can aid debugging, but will doing so also block secure media? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP:

Is DTLS available for Centos? Either Centos 4 or 5. DTLS is TLS over UDP. Highly valued to protect SIP traffic.....

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

I've got a Zultys WIP2 and Zultys 2x2 both of which support encryption. I have patched my asterisk with srtp (srtp.sourceforce.net) as well as with the patch found at http://bugs.digium.com/view.php?id=5413. I'm trying to utilizing the encryption feature of the two Zultys phones to create an encrypted call, but am having extreme difficulty. I keep getting the following error message

Hi all, I am looking for a secured communication between web clients and my servers. tinc looks great. I understand it uses UDP for data. But does it use DTLS (newbbie question) ? As someone tryed to use 0MQ with it ? Cheers, Laurent. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I get the following error when using our Let's Encrypt ssl certificate for webRTC calls : [Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled [Jun 2 14:29:28] ERROR[27360][C-00000ae5]: res_rtp_asterisk.c:1441 ast_rtp_dtls_set_configuration: Specified certificate file '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance

I'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

Hi Everyone. I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer >From the sdp can anyone suggest why secure audio cannot be provided ????v=0 ????o=- 6611325078116277019 2 IN IP4 127.0.0.1 ????s=- ????t=0 0 ????a=group:BUNDLE audio ????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l ????m=audio

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hi all, i try to deactivate SRTP in asterisk 11. In sip.conf: tlsenable=no encryption=no transport=udp srtpcapable=no but when I try to make a call comes following message: [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1

Hi, I am using chrome version 36 and opera with asterisk 11.9.0 and cent os I am getting the below error if i do call on sipml5 from blink 1. Failed to set remote offer sdp: Called with SDP without DTLS fingerprint. tsk_utils.js?svn=224:128 1. tsk_utils_log_errortsk_utils.js?svn=224:128 2. tmedia_session_jsep01.onSetRemoteDescriptionError

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the

The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs