search for: res_srtp

Hello The "res_srtp" module does not appear. How do I install it? Thanks.

Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has encrytion->yes The client program is CSipSimple on Android H...

...ll with "yum install libsrtp-devel") > > Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it enabled. It's okay if pjproject doesn't as we don't use their media layer. Do you have the res_srtp module in Asterisk? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

...SIP RTP CoS mark 5 -- Called SIP/10000 -- SIP/10000-00000013 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [10000 at default-outbound08:2] VoiceMail("SIP/10002-00000012", "10000,uj") in new stack [Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure [Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure -- <SIP/10002-00000012> Playing 'vm-theperson.g729' (language 'en') -- <SIP/10002-00000012>...

Hi, I'm getting messages like res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 because of authentication failure 10 == SRTP unprotect failed on SSRC 576693764 because of authentication failure 160 [...] ... after a couple minutes...

Hi; It seems the SNOM Phones are requesting to have SRTP but I do not have the module res_srtp. I tried to compile it with asterisk 1.8, make menuselect, but I found that it can not be used (I am not able to select it) with the following details: Secure RTP SRTP Depends on: srtp E Can use: N/A Conflicts with: N/A So, how I can use it? What I have to do to know the reason for not being ab...

....6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/04e5bee5/attachmen...

...tps://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. TLS is fine and I can call between clients. SRTP is a different matter, my SIP clients return: SIP 488 "Not acceptable Here" I'm really stumped on this one, any ideas? srtp module is loaded: *CLI> module show like res_srtp.so Module Description Use Count res_srtp.so Secure RTP (SRTP) 0 1 modules loaded extensions.conf extract exten => 1002,1,Set(_SIPSRTP=${SIPPEER(1002,srtpcapable)}) exten => 1002,n,Set(CHA...

On Thu, Jan 16, 2020 at 11:35 AM hw <hw at gc-24.de> wrote: > On Tuesday, January 14, 2020 5:29:04 PM CET hw wrote: > > Hi, > > > > I'm getting messages like > > > > > > res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay > check > > failed (index too old), retrying == SRTP unprotect failed on SSRC > 576693764 > > because of authentication failure 10 == SRTP unprotect failed on SSRC > > 576693764 because of authentication failure 160...

Hi folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long

...000033) > == Spawn extension (local, 301, 1) exited non-zero on > 'SIP/hsolutionspf5-00000002' I tried many options to disable SRTP but without success : * canreinvite = no * canreinvite = nonat * srtpcapable=no * encryption=no * directmedia=nonat * ...or noload => res_srtp.so in modules.conf Any help would be GREATLY appreciated ! Denis P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/3bb51d88/a...

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

...any answers that would solve my case so I must turn to you guys. In my sip.conf I have savpf=yes, but is there something else I need to enable or change in the configs or change my peer configurations? I'm not sure if this is relevant but I checked that Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes ; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain...

...IP/sip_channel_params --- PASSED --> tests/channels/SIP/sip_tls_call --- FAILED --> tests/channels/SIP/sip_tls_register --- PASSED --> tests/channels/SIP/sip_srtp --- SKIPPED --> Dependency: twisted -- Met: True --> Dependency: starpy -- Met: True --> Dependency: res_srtp -- Met: False --> tests/channels/SIP/noload_res_srtp --- PASSED --> tests/channels/SIP/noload_res_srtp_attempt_srtp --- SKIPPED --> Dependency: twisted -- Met: True --> Dependency: starpy -- Met: True --> Dependency: res_srtp -- Met: False --> tests/channels/SIP/...

...350> (type 03, len 000033) == Spawn extension (local, 301, 1) exited non-zero on 'SIP/hsolutionspf5-00000002' I tried many options to disable SRTP but without success : * canreinvite = no * canreinvite = nonat * srtpcapable=no * encryption=no * directmedia=nonat * ...or noload => res_srtp.so in modules.conf Any help would be GREATLY appreciated ! Denis P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Joi...

On Tuesday, January 14, 2020 5:29:04 PM CET hw wrote: > Hi, > > I'm getting messages like > > > res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check > failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 > because of authentication failure 10 == SRTP unprotect failed on SSRC > 576693764 because of authentication failure 160 [...] > > > ....

...t be secure. The Asterisk_SRTP_2 box is registered in the Asterisk_SRTP_1 box as it was a client. The problem is that, when i call from phone1 to phone2, when phone2 picks up, the call is automatically hungup by Asterisk_SRTP_1 with the following error message: [May 2 17:10:33] DEBUG[11553]: res_srtp.c:456 res_srtp_unprotect: SRTP unprotect: authentication failure [May 2 17:10:33] WARNING[11553]: rtp.c:1648 ast_rtp_read: RTP Read error: Success. Hanging up. Is this a known error? What could be the problem here? Here are the important configuration files: - Asterisk_SRTP_1: sip.conf:...

Apologies in advance if this has come up a thousand times before but is there any way to stop this error in 1.8 ? [ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup SRTP session. -- Thanks, Phil

...Joseph) * ASTERISK-23509 - [patch]SayNumber for Polish language tries to play empty files for numbers divisible by 100 (Reported by zvision) * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-24502 - Build fails when dev-mode, dont optimize and coverage are enabled (Reported by Corey Farrell) * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp) * ASTERISK-260...

...3 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24432 - Install refcounter...