Freepbx version 4.211.64-7
Asterisk version 11.5.1
Whenever I try to complete a call using WebRTC, call stops as soon as I hit send, and asterisk give me this message:
[2014-05-30 12:25:03] WARNING[C-00000440]: chan_sip.c:10454 process_sdp: Rejecting secure audio stream without encryption details: audio 2363 RTP/SAVPF 111 103 104 0 8 106 105 13 126
I’m guessing this means I need to provide some sort of authentication but I don’t see where. Thanks in advance for any help.
Here’s the thing about WebRTC. The browsers and standards change weekly. If you think Schmooze has taken the standpoint of not wanting to add user & device support or that I am ignoring the tickets its because this is changing so rapidly that its not worth it to invest a ton of time into it. Even the Asterisk developers don’t invest much time. We are all waiting for everything to settle down.
I have seen the error you mentioned and I have no solution for you. The code is the same code it was last August when I wrote it, there is nothing different and it worked then but now it’s not working for a large majority of users and it’s not something I am actively looking at, at the moment. Sorry for brutal honestly but I wouldn’t say WebRTC is or will be reliable for another two years. Right now it’s just a play toy, if you get it to work good for you otherwise I am sorry.
I never said anything about ignoring tickets or Schmooze taking a negative standpoint on anything. I was merely asking for any help or advice if anyone had it working and had a minute to spare to help me out. If not, that is fine and I can move on to another solution, but there is no need to take out any frustrations you may have from other people badgering you on me. I seem to remember seeing a post stating that no one is forcing people to post a response on these forums. Thanks to everyone who tries to help the less experienced on here.
I never took out any frustrations on you. You are looking way too deeply into my post. I was posting generally so that I can reference this thread when this comes up again. I am the one who also wrote about “no one is forcing you to post”. I wrote WebRTC for this company so I thought you’d want a reply directly from the source. No one else would have an answer for you.
Sorry if I have offended you but I think you completely misunderstood what I was trying to convey.
I am locking this thread so it doesn’t turn into a finger pointing game. I’ve already gotten a call from my boss whom your boss called about how I treated you in this thread so I think everything in this thread has been taken care of. Have a good day.
Just to chime in on here: this is not a FreePBX problem, it is an Asterisk problem; or at least, an Asterisk/browser compatibility problem.
When the IETF made DTLS-SRTP as mandatory to implement and SDES-SRTP as “shall not implement”, the browsers migrated themselves over to DTLS. Eventually - in Chrome 35 (I think, I’m unsure of the Firefox version) - they stopped offering SDES completely. That’s not a bad thing, it’s a good thing. DTLS-SRTP is a lot more secure than SDES-SRTP, and it is a good decision for the future of WebRTC. Unfortunately, both Firefox and Chrome negotiate DTLS-SRTP slightly differently, and neither are compatible with Asterisk’s implementation of DTLS-SRTP.
Asterisk has supported DTLS-SRTP since Asterisk 11 was first released. However, when we wrote DTLS support for Asterisk, almost no one else out there supported it - except for a few proprietary smart phones that didn’t want to play nice with anyone. Since there was no one to test with, we made sure that Asterisk could pass DTLS encrypted media back and forth between Asterisk instances. Unfortunately, we didn’t implement support for SHA-256, which Chrome (at the very least) requires. Hence, the problems with recent versions of browsers and Asterisk.
There is an open issue in the Asterisk issue tracker for this problem - https://issues.asterisk.org/jira/browse/ASTERISK-22961. There are some preliminary patches on the issue for Asterisk 11 - but developers are still working on them. If you’d like to help out with testing the patches, that would be appreciated.
But until the Asterisk project gets this straightened out, there’s not much the folks here on the FreePBX project can do