similar to: 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

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:

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 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

If I understand correctly, setting encryption=no means that Asterisk will make outgoing calls without 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

Hi all, I''m a newbbie in this kind of things. I need an example to configure QOS over a WAN for http, telnet and ftp or a web-interface aplication to start working with QOS. (such as webmin with iptables, but for tc). Thanks in advance Alem.

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, 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

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

Running CentOS release 5.5. I'm trying to update or patch an SRPMS file, specifically openssl-0.9.8e-12.el5_4.6.src.rpm. Basically, I'm trying to change one line in the source, in ssl/ssl.h. I create a "diff ?u" file called openssl-ssl-h.patch. I then edit the openssl.spec file, and add 2 lines to that in the appropriate place: Patch88: openssl-ssl-h.patch And

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

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

To wit: OpenSSL Security Advisory [9 Jul 2015] ======================================= Alternative chains certificate forgery (CVE-2015-1793) ====================================================== Severity: High During certificate verification, OpenSSL (starting from version 1.0.1n and 1.0.2b) will attempt to find an alternative certificate chain if the first attempt to build such a chain

The Asterisk Development Team would like to announce security release Asterisk 21.0.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside

The Asterisk Development Team would like to announce security release Asterisk 21.0.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside

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

The Asterisk Development Team has announced the release of Asterisk 13.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.10.0 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:

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from

hello, my webrtc calls ends after ~60seconds with "res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating". any ideas where can be problem? or howto debug this problem? asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox) -- --------------------------------------- Marek Cervenka