search for: dtls

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 attachment wa...

...ing directmedia=no ; Asterisk will relay media for this peer transport=ws ; Asterisk will allow this peer to register on UDP or WebSockets force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 nat=force_rport,comedia accept_outofcall_message=yes outofcall_message_context=messages ;dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer ;dtlsverify=no ; Tell Asterisk to not verify your DTLS certs ;dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is ;dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key...

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 and Chrome on both Windows 8.1 x86 and Android 4.3), I can successfully establish a call but I can hear NO SOUND. As it turns out (according to wireshark logs), Asterick is sending me DT...

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> 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? >

...ng 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 regards, Ruban.S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://l...

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: <http://www.tinc-vpn.org/pipermail/tinc/attachments/20130517/d28319c9/attachment.html>

...hout encryption, but will be happy to accept incoming calls regardless of whether the caller wants encryption or not If encryption=yes, then Asterisk not only uses encryption for the outgoing calls but it will refuse to accept incoming calls unless they use encryption too If I have encryption=no dtlsenable=yes the DTLS support works but Asterisk will no longer accept incoming calls using regular RTP/AVP. These messages appear in the console and the call is rejected with code 488: [Jan 28 11:08:42] WARNING[24673][C-00000009]: chan_sip.c:10496 process_sdp: Processed DTLS [FALSE] [Jan 28 11:08:...

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: 0x997A9F17ED7DAEA6

...g, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). is there any way to configure to have the previous behaviour? Im trying to set dtlscipher=AES128-SHA but I always see DTLS ECDH initialized (automatic), faster PFS enabled any idea? Thanks! res_rtp_asterisk ------------------ * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS). Enabling PFS is attempted by default, and is dependent on the configuration of the...

Hello list, I have successfully set up Asterisk, but girls from our office complain to me that when they hit Flash to transfer a call and pick the number, they need to wait until the call is answered, and only then they could hangup. On the analog PBX we had before the transfer was in "background", and if called party did not answer the call, then the call returned to the girl in the

...registrations support WS or WSS as valid transports (not true) (Reported by PSDK) * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov) * ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handsh...

...registrations support WS or WSS as valid transports (not true) (Reported by PSDK) * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov) * ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handsh...

...nes: Rejecting secure audio stream without encryption details: audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 Trying to put 'SIP/2.0 488' onto WS socket destined for www.xxx.yyy.zzz:5060 No compatible codecs for this SIP call. Here is the sip.conf info. I have tried various permutations of the dtls and encryption parameters with no luck. I do have openssl and srtp built into Asterisk (that solved a different error dealing with the RTP engine). [webrtc-dtls] ; Add DTLS stuff for Mozilla Nightly (and eventually Firefox) type=user host=dynamic hassip=yes transport=ws,wss dire...

Hi All We've been reading this in the CLI a lot lately: Received SSL traffic on RTP instance '0x7fe7481faad8' without an SSL session How can we find details about this particular RTP instance? "rtp set debug" needs an IP which is precisely what I want to know (and I don't)! Cheers Ethy

...registrations support WS or WSS as valid transports (not true) (Reported by PSDK) * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov) * ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handsh...

Hey, RELENG_7 isn't -STABLE yet, so the issue mention in the commit mail beolow will not get a Security Advisory. This only affects applications using DTLS, and I doubt there are many of those, but users should still upgrade to get this fix, just in case. See the OpenSSL advisory for some more details: http://www.openssl.org/news/secadv_20071012.txt If anybody were wondering, and hadn't checked the OpenSSL advisory: older versions of FreeBSD are...

Hey, RELENG_7 isn't -STABLE yet, so the issue mention in the commit mail beolow will not get a Security Advisory. This only affects applications using DTLS, and I doubt there are many of those, but users should still upgrade to get this fix, just in case. See the OpenSSL advisory for some more details: http://www.openssl.org/news/secadv_20071012.txt If anybody were wondering, and hadn't checked the OpenSSL advisory: older versions of FreeBSD are...

...6 (%prep) The ssl.h.rej file has: *************** *** 497,503 **** /* SSL_OP_ALL: various bug workarounds that should be rather harmless. * This used to be 0x000FFFFFL before 0.9.7. */ - #define SSL_OP_ALL 0x00000FFFL /* DTLS options */ #define SSL_OP_NO_QUERY_MTU 0x00001000L --- 497,503 ---- /* SSL_OP_ALL: various bug workarounds that should be rather harmless. * This used to be 0x000FFFFFL before 0.9.7. */ + #define SSL_OP_ALL (0x000...

...hat was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully started when completing attended transfer (Reported by Joshua Colp) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov) * ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handsh...