similar to: Asterisk SIP and SRTP

Hello all, I want to use Asterisk with the SRTP patch from http://bugs.digium.com/view.php?id=5413 . I'm confused to create the certificates for it. Can anybody help in such question? P. S. I've created the pem files and renamed it to * ${astetcdir}/asterisk.crt * ${astetcdir}/asterisk.key * ${astetcdir}/ca-certificates.crt but the asterisk got "segmentation fault" error at

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>

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, 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

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 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

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

Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). ---------snip------------------ == Using SIP RTP CoS mark 5 -- Executing [10000 at

On 2/23/19 2:39 PM, Social Boh wrote: > *DIrect media with SRTP is not supported. All media when SRTP goes > through Asterisk.* > > So you have to open ports on your firewall and disable directmedia=yes > on your configuration. directmedia is not explicitly enabled; I guess it's the default. Joshua basically says there is no way to control which ports are being used for

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

On 2/23/19 4:19 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 11:04 AM, hw wrote: > > <snip> > >> >> directmedia is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >>

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'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

Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said: ============================================== *Note: There is

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

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:

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

One of my clients today asked me about TLS support for encryption of SIP payloads, and I didn't have an adequate answer as to why it wasn't supported or even discussed. Some archive searching finds scant mention of this in reference to Asterisk. Of course, encrypting the SIP payload is only 1/2 the problem; the payload itself is the next problem. I understand that IAX solves these

Hi. I have an Asterisk Server (A) where it acts as the main gateway to offer services. There are different asterisk servers (B -D) that connect as extensions to the Server A. I would like to implement TLS and SRTP for these extensions, but have the non secure as well for other extensions. for example the extensions 4500-4504 be with TLS/SRTP and the rest be non secure(ordinary). Is there a guide