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| 22:16.33 | P-NuT | Hi All, I have put together an asterisk 13 server on Debian 7 from source. |
| 22:18.08 | P-NuT | When I configure SIP/TLS registrations, and register an endpoint it works fine. When I call without srtp it also works fine. |
| 22:18.49 | P-NuT | However, when I call using SRTP I get the following error: Rejecting secure audio stream without encryption details: audio 4002 RTP/SAVP |
| 22:19.09 | P-NuT | Can anyone point me in the right direction? |
| 22:20.50 | mjordan | do you have encryption details (that is, a crypto line) in the SDP? |
| 22:21.10 | P-NuT | What info can I give you? |
| 22:21.32 | P-NuT | Actually, I can paste that line for you, just one sec. |
| 22:21.39 | mjordan | pb your SDP |
| 22:22.01 | mjordan | and this is really appropriate for #asterisk |
| 22:27.16 | P-NuT | I've tried in asterisk, they have no idea. |
| 22:27.18 | P-NuT | http://pastebin.com/nUKxDcBi |
| 22:30.17 | P-NuT | What do you think? |
| 22:38.28 | mjordan | what's your SIp peer configuration? |
| 22:39.23 | P-NuT | I have a config in pjsip.conf and one in sip.conf. |
| 22:39.34 | P-NuT | I have followed this tutorial |
| 22:39.36 | mjordan | well, this is sip.conf |
| 22:39.47 | P-NuT | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 22:40.12 | mjordan | Are both SIP channel drivers bound to different addresses? |
| 22:40.16 | mjordan | And are you actually using both? |
| 22:40.32 | P-NuT | it would look that way |
| 22:40.41 | mjordan | I can't tell |
| 22:40.49 | mjordan | If you have both bound to 5060, weird things will happen |
| 22:41.06 | mjordan | make sure that if you are using both, and both are loaded, that they are listening on different addresses. That may not fix your issue, but you should double check that |
| 22:41.37 | P-NuT | So, a bit of background, I have come from 1.8 to 13, so this whole pjsip thing and describing peers as endpoints is completley foreign to me. |
| 22:42.24 | P-NuT | So, I understand chansip as that was in 1.8, but have no idea about pjsip. Can I disable it completley and use it like I did with 1.8? |
| 22:42.47 | mjordan | it's two separate channel drivers. |
| 22:42.50 | mjordan | You can use either. |
| 22:43.00 | mjordan | You can use both, but if you do, you have to make sure they aren't trying to both service the same SIP requests |
| 22:43.27 | P-NuT | So how do I disable pjsip entirely to ensure that I am only using chansip? |
| 22:43.36 | mjordan | the same way you would for any other module |
| 22:43.38 | mjordan | don't load it |
| 22:43.41 | mjordan | or don't build it |
| 22:43.58 | mjordan | or tell it to bind to a different address |
| 22:43.58 | P-NuT | ok, let me try that. Can I then define the user in users.conf? |
| 22:44.03 | mjordan | er |
| 22:44.08 | mjordan | please don't? |
| 22:44.15 | mjordan | users.conf is not supported in any way. |
| 22:44.22 | mjordan | your mileage will vary. |
| 22:44.34 | P-NuT | ok, so define the user in sip.conf? |
| 22:44.42 | mjordan | it's not a user |
| 22:44.49 | mjordan | it's a peer/friend in sip.conf |
| 22:45.16 | P-NuT | ok, so lets call it a peer. |
| 22:45.37 | mjordan | that's what it is, so sure. Terminology matters when conveying information :-) |
| 22:47.05 | P-NuT | Thanks, sorry I'm so vague. |
| 22:47.30 | P-NuT | So, from the console if I run modules show I dont see anythng saying pjsip |
| 22:48.45 | P-NuT | can you tell me what the modukle names are? |
| 22:49.08 | mjordan | P-NuT: you can find that out yourself |
| 22:49.30 | mjordan | look for anything named 'pjsip' in channels/ and res/ |
| 22:49.49 | P-NuT | ok |
| 22:52.53 | P-NuT | well, if I run module show |
| 22:53.28 | P-NuT | I dont see anything saying pjsip, so I assume it's not loaded. |
| 22:54.40 | mjordan | that would be the case then |
| 22:55.29 | P-NuT | ok cool, so does that mean that It wont read pjsip.conf? |
| 22:56.30 | mjordan | nope. So you only need to pastebin your sip.conf. |
| 22:59.48 | P-NuT | brilliant. |
| 22:59.53 | P-NuT | Right, let me do that |
| 23:04.53 | P-NuT | http://pastebin.com/mS6k6bXE |
| 23:07.37 | P-NuT | Any clues? |
| 23:09.34 | mjordan | you need to set the encryption option |
| 23:09.42 | P-NuT | Ok.. |
| 23:10.01 | mjordan | your peer is 12341234, right? |
| 23:10.07 | mjordan | add encryption=yes under it. |
| 23:11.25 | P-NuT | yes, I have just doen that and then done a sip reload. |
| 23:11.29 | P-NuT | I get the same error |
| 23:11.47 | mjordan | k. You still need that. |
| 23:16.37 | P-NuT | the encryption=yes line? Yes I get that. But again after a SIP reload I get the exact same error. It's a really puzzling one and all the innfo I saw online looked like it was something to do with webrtc. |
| 23:24.29 | P-NuT | Any ideas? |
| 23:30.15 | mjordan | nope. |
| 23:30.20 | mjordan | At this point, you'd need to get a full debug log. |
| 23:30.31 | mjordan | That would illustrate what Asterisk is doing when it interprets the SDP. |
| 23:30.50 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:41.54 | P-NuT | AAhhhhhhhhhh. When looking at the debug I see this.. No SRTP module loaded, can't setup SRTP session. |
| 23:53.19 | mjordan | you need libsrtp and res_srtp |
| 23:53.32 | mjordan | libsrtp can be found on github (and you'll want to use the version from there) |
| 23:53.39 | mjordan | res_srtp can be built once that is installed |
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