If I understand correctly,Asterisk 14 introduced support for some new SRTP ciphers (including some 256 bit ones), previously only two 128 bit ciphers were supported. Using Asterisk 14, I was able to make a call from a softphone (Groundwire) with a ..
I have a PJSIP account configured as below. I am testing with the Echo Test application on Asterisk 13 and using CSipSimple.I can create a call with TLS and SRTP, however for some reason only 1 in every 5 calls has audio.When I connect over WiFi, I h..
,I have a very strange problem :* external calls work perfectly,* internal calls between some phones too,* but internal call between two similar phones dont work !!! (Snom 710)When we have sound, there are no errors in asterisk. When we do not have sou..
Does anyone here have a Jitsi softphone set up with Asterisk such that SRTP is enabled, TLS is used to pass the SRTP key, and it works?Anyone?If so then what are the settings required for..
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 exam..
Ive 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.Than..
I have a question about best practice (or recommended practice) for allowing SIP registrations from the Internet. This is what I was thinking of implementing:1. Use OpenSips for the SBC,enable SRTP and TLS2. Allow limited access to the actual Aster..
all,Im 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-15381That indicated that Asterisk doesnt support TLS as an OPTIONAL transport. Its eit..
Ive just done a test with a WebRTC client connecting to the repro proxy with the SIP messages relayed over TCP to AsteriskAsterisk successfully answers the call using SAVPF, SRTP and ICE.The client is greeted by the demoThis was tested in the Aster..
I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.orgI followed this guide:https://wiki.asterisk.org/wiki/display/AST/Asterisk+PackagesThe SRTP support appears to be missing though.I notice libsrtp was not automatically instal..