Unsolved: Solved: How To Create A Working Certificate For Using TLS?
For easybell, I don’t know of any advantage. But that’s not very reliable, because I’m using easybell for dedicated requirements only. I’m considering chan_sip legacy. I wouldn’t build any new installation on chan_sip (if there are no technical contradictions).
Easybell does have a pretty fine documentation for FreePBX and pjsip:
https://www.easybell.de/nc/hilfe/ergebnis/freepbx-130124-mit-asterisk-13.html
[why encryption?]
That’s true. But don’t forget to encrypt RTP at this point! This must be done additionally. BTW: easybell doesn’t support full RTP encryption. It’s supported for outgoing calls only
(https://en.easybell.de/nc/help/questions/questions-concerning-the-telephone-connection/answer/does-easybell-support-the-data-encryption-srtp-for-voip.html).
That’s an example for an inbound call – there isn’t any support for RTP encryption:
<--- Received SIP request (869 bytes) from TLS:195.185.37.60:5061 --->
INVITE sip:+4912345678989@93.165.22.128:5061;transport=TLS;line=xxxxxxx SIP/2.0
Via: SIP/2.0/TLS 195.185.37.60:5061;branch=z9hG4bKcu8ZTaf6c4iPU;rport From:
To:
CSeq: 1891 INVITE
Call-ID: SBC4115e0000af5-5d21b654-67c091e3-a60cca0-924117e-01_b2b-1
Max-Forwards: 68
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK
Supported: histinfo Content-Type: application/sdp Content-Length: 286
Contact:
v=0
o=- 1667528048 2824605765 IN IP4 195.185.37.60
s=-
c=IN IP4 195.185.37.60
t=0 0
m=audio 30934 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:both a=sendrecv a=maxptime:150
True – no contradiction of mine.
Happy to here that it’s working now.
There are some config changes which need a complete restart AFAIK (FreePBX explicitly warns about some changes requiring a restart).
That’s how I’m handling it: After each change concerning the transport I’m restarting asterisk. Just to be sure. But that’s a question which could be answered by Joshua much better.
The certificate check is done on starting the connection (SYN) by openssl. sip reload most probably doesn’t restart the connection (because all running calls would be disconnected – that’s most of the time not a good idea – sip reload usually doesn’t destroy running sessions / calls).
Just handle it like this: each transport relevant change requires a complete restart!
I think it’s supposed to behave like this, because it would mean to disconnect all running calls on sip reload. That’s probably not what most of the people expect / want to have.
Regards Michael
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2 thoughts on - Unsolved: Solved: How To Create A Working Certificate For Using TLS?
Yes, easybell works fine, and their support is great. But don’t tell anyone or they might be overwhelmed with customers fleeing the bad support of other providers …
Is there an advantage to using pjsip? What’s needed for easybell with pjsip?
I consider it a requirement for when employees end up using their mobile phones over foreign wireless networks, which is something that would be virtually impossible to prevent should the asterisk server be made reachable from the outside.
And before that, why shouldn’t phone calls always be encrypted for just in case? They are always genuinely private, and it doesn’t hurt anything.
Thanks, setting ‘tlscafile=/etc/pki/tls/certs/ca-bundle.crt’ seems to do the trick. However:
First I set ‘tlsdontverifyserver=no’ and issued a ‘sip reload’. There was no error message. I found that suspicious and restarted asterisk, and the error message came back.
Only then I added ‘tlscafile=/etc/pki/tls/certs/ca-bundle.crt’ (which was unset before), and after a ‘sip reload’, the error message was gone. So far, it hasn’t come back even when restarting asterisk.
This shows that ‘sip reload’ doesn’t really do a reload in that a certificate which hasn’t been verified continues to be accepted after the configuration changed to now require verifying the certificate. This might be a security problem, and if not, it is certainly good for surprises and can create much confusion.
Is it supposed to be like this, or should I make a bug report?
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That’s not for asterisk, and most documentations for asterisk are not for pjsip.
They also say that encryption is possible:
https://en.easybell.de/nc/help/questions/questions-concerning-the-telephone-connection/answer/can-i-encrypt-the-telephony.html
I’ll have to see what their support says about this.
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That would really suck. It is not acceptable that phone calls over the internet shouldn’t be encrypted.
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