similar to: snom and srtp

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 during voice calls after which the connection is being

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 > >

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

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm running Asterisk 11.3.0 on wheezy. I'm trying to do TLS +SRTP with blink SIP clients as shown here https://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

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 able to

Hi all, I upgraded one of our servers running asterisk 1.6.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

Hi! I am having trouble getting the following configuration to work: PHONE1 <-- rtp --> Asterisk <--IAX--> Asterisk_SRTP_1 <--- srtp ---> Asterisk_SRTP_2 <-- rtp--> PHONE2 This means, I am using regular voip clients without srtp support on both sides, but the communication between the 2 Asterisk_SRTP boxes must be secure. The Asterisk_SRTP_2 box is registered in the

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

Jonas Kellens wrote: > Question : I noticed I received an error when installing pjproject > --with-external-srtp > > I do not seems to have the srtp capability. > (However I can easily install 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

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

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

On 2/22/19 7:56 PM, Joshua C. Colp wrote: > On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: >> >> Hi, >> >> when trying to use SRTP, I can see UDP traffic from phones to the >> asterisk server being dropped be the firewall on arbitrary ports. > > There is no separate port range used for SRTP, and Asterisk does not control the port that the phone uses for sending

Hi experts, I am trying Asterisk SRTP in my environment, and find that when Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay by Asterisk are local ports on the Asterisk server, media from the two clients out of the NAT (for example from Internet) can not reach the ports, and thus the two client can not establish the secure call via Asterisk. I have set up a STUN server

On 2/23/19 1:15 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 8:06 AM, hw wrote: >> On 2/22/19 7:56 PM, Joshua C. Colp wrote: >>> On Fri, Feb 22, 2019, at 2:48 PM, hw wrote: >>>> >>>> Hi, >>>> >>>> when trying to use SRTP, I can see UDP traffic from phones to the >>>> asterisk server being dropped be the firewall

Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? That the traffic is being blocked by the firewall is probably the reason

> Olle E Johansson wrote: >> >> 23 apr 2007 kl. 19.55 skrev Russell Bryant: >> >>> John Todd wrote: >>>> To morph this into a -dev thread: if this patch were to become (again) >>>> useful and error-free, is there any objection or usefulness in adding it >>>> to TRUNK? Personally, I think there is, if there is a method by which

I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module exists under /usr/lib64/asterisk/modules Is it still necessary to do a source build every time SRTP is

Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. Thanks, Patrick