search for: libsrtp

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for...

I've been having some issues with srtp. so I checked which version of libsrtp I built asterisk 11.6 against. I'm on fedora 21, so libsrtp-1.4.4-13.20101004cvs.fc21.x86_64. From https://github.com/cisco/libsrtp it seems that latest release is 1.5.1, released a couple of weeks ago. I'm not a fan of the bleeding edge, but using a version 4+ years old seems strange...

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure: WARNING: *** replacing /usr with the prefix of your choice. configure: WARNING: *** After re-installing li...

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 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? --...

Hello, I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. I followed this: cd /usr/src wget ... asterisk-13.13.1.tar.gz tar zxf asterisk-13.13.1.tar.gz cd asterisk-13.13.1 ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr" ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled make menuselect (shows res-srtp is available) make latest make command fails with

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 all, can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 on it with options as stated below - ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ --with-pjproject-bundled when I tried to run "make menuselect". i get the error below. Makefile:109: makeopts: No such file or directory **** **** The configure script must be executed

...packets with invalid UPD checksums, then the packages must be somehow altered and their UPD checksums recalculated to arrive here. Does bad hardware etc. do that? Why would the UDP checksums just happen to get recalculated correctly but like randomly without intent? Only when asterisk (i. e. libsrtp) finally verifies the authentication tag of an SRTP package against the authenticated part of the package --- which, according to RFC 3711, seems to be the entire payload of the UPD package --- the verfication fails. How is that possible?

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 needed? Or is the srtp module distributed in some other rpm?

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

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

Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com

...P being interrupted at irregular intervals. The intervals appear to depend on the time of day: Such phone calls can last for a duration of about 5--25 minutes during the day to up to 1.5 hours at around 3am before being interrupted. Asterisk says that a package is being replayed, meaning that libsrtp has already seen and processed the packet earlier. That can happen a couple times until asterisk reports authentication failures. The result is that the call is interrupted in that I can not hear the opposite end while the other end sometimes can still hear me, sometimes not. The interruptio...

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

The Asterisk Development Team has announced the release of Certified Asterisk 13.1-cert6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.1-cert6 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas

...how busy the routers/firewalls between 2 users can be. How would packets being dropped explain the replay errors and authentication failures? > Packets can get > out of order or a dozen other things which then relies on the > application layer to put the things back in 'order'. libsrtp seems to have provisions to deal with packets arriving out of order. > For voice, that > usually means a drop or other ugliness because it is assumed that if > the quality is too bad, the people would just call each other again. That's a funny idea. Phone calls just worked fine and...

...hat you are asking. But it's unlikely (very unlikely) that the checksums are randomly correct. But packet checksums are recalculated when packets are forwarded by layer 4 switches - the contents of the package are inspected as part of the switching process. > > Only when asterisk (i. e. libsrtp) finally verifies the authentication tag of > an SRTP package against the authenticated part of the package --- which, > according to RFC 3711, seems to be the entire payload of the UPD package --- > the verfication fails. > > How is that possible? > If it's SRTP checksum...

Hi there! My issue is that i have (bouht to me) a box with Asterisk 1.8.2.2 but its seems that with no SRTP support. So i added a libsrtp libraries and like i understand now i need to recompile/reinstall Asterisk... is that safe to all data and MySQL backand data? -- Best regards Matiss Jekabsons Procerto Ltd. Dzelzavas Str. 117. Riga, Latvia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://...

...umber 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-26034 - T.38 passthrough problem behind fi...