[asterisk-bugs] [JIRA] (ASTERISK-22675) Asterisk refuses correct RTP/AVP with optional encryption

[Dmitriy Serov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22675?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211168#comment-211168 ] 

Dmitriy Serov commented on ASTERISK-22675:
------------------------------------------

I hope patch from [issue ASTERISK-22005|https://issues.asterisk.org/jira/browse/ASTERISK-22005] can help.
And it helps: outgoing call with forced "on" client (CSipSimple) SRTP and encryption=no is successefull!
BUT: But incoming calls rejected with cause 58 and warning "chan_sip.c:10512 process_sdp: We are requesting SRTP for audio, but they responded without it!"
I still don't have optional encryption :(
                
> Asterisk refuses correct RTP/AVP with optional encryption
> ---------------------------------------------------------
>
>                 Key: ASTERISK-22675
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22675
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 11.5.1
>            Reporter: klaus3000
>
> Asterisk receives an INVITE from Blink:
> {noformat}
> INVITE sip:echo at 83.136.32.165:4343 SIP/2.0
> Record-Route: <sip:83.136.32.159;lr=on;ftag=a25a6db049c34fa5bb8805dee3d46aa2;relay=yes;nat=caller>
> Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK4a92.27b2ea15.0
> Via: SIP/2.0/UDP 198.19.188.196:62329;received=63.133.202.2;rport=62329;branch=z9hG4bKPj0f5d2851846d46db9520756755856edc
> Max-Forwards: 16
> From: "Klaus Darilion" <sip:klaus.darilion at labs.nic.at>;tag=a25a6db049c34fa5bb8805dee3d46aa2
> To: <sip:8001 at labs.nic.at>
> Contact: <sip:klaus.darilion at labs.nic.at;alias=63.133.202.2~62329~1;gr=urn:uuid:2847198f-3fd0-421d-9619-50c2e744af33>
> Call-ID: c46dde8cba7044539c84829ea6975f96
> CSeq: 16406 INVITE
> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
> Supported: 100rel, replaces, norefersub, gruu
> User-Agent: Blink 0.5.0 (Windows)
> Content-Type: application/sdp
> Content-Length: 579
> X-Info: (83.136.32.159:30067) Enforcing NAT traversal for local caller
> X-Info: (83.136.32.159:30067) Enforcing symmetric response routing
> v=0
> o=- 3590409414 3590409414 IN IP4 83.136.32.159
> s=Blink 0.5.0 (Windows)
> c=IN IP4 198.19.188.196
> t=0 0
> m=audio 11000 RTP/AVP 108 99 98 9 0 8 96
> c=IN IP4 83.136.32.159
> a=rtcp:11001
> a=rtpmap:108 opus/48000
> a=rtpmap:99 speex/32000
> a=rtpmap:98 speex/16000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-15
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y3I9RSOpehl6UG/pMX1CaNeSak1cKOocNdElKqXX
> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:ZIfBn8Ph00X7eEgD9Y/Ixaj0Kjgv3SiFnlPlqi9E
> a=sendrecv
> a=nortpproxy:yes
> {noformat}
> The SDP uses the RTP/AVP profile, but adds a=crypto lines. This means: "RTP is fine, but if you also support SRTP when can make SRTP too".
> Regardless of the "encryption=..." setting, Asterisk refuses this call with:
> "We are requesting SRTP for audio, but they responded without it!"
> Asterisk should accept the call and use RTP if encryption=no or use SRTP if encryption=yes.
> Even better would be if encryption= also allows "yes, if possible". This would allow SRTP calls with fallback to RTP.

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