[ https://issues.asterisk.org/jira/browse/ASTERISK-20234?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=195822#comment-195822 ]
Matt Jordan commented on ASTERISK-20234:
----------------------------------------
We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Please make sure that 'sip set debug on' is enabled as well.
> SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
> ----------------------------------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-20234
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20234
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/SRTP
> Affects Versions: 10.7.0
> Environment: RHEL5 linux 2.6.18-308.11.1.el5
> Reporter: tootai
>
> As you can see, snom 320 (latest stable firmware snom320-SIP 8.7.3.10) is annoncing crypto but asterisk doesn't recognize it.
> v=0
> o=root 80443371 80443371 IN IP4 192.168.10.105
> s=call
> c=IN IP4 192.168.10.105
> t=0 0
> m=audio 49154 RTP/AVP 9 0 8 3 99 108 18 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:nCj5W+omFyJ2qyncFHRWrUBcmSSOVXAs7E9xQy7x ;<- here
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> <------------->
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: --- (20 headers 19 lines) ---
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Sending to xxx.xxx.xxx.xxx:2048 (NAT)
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Using INVITE request as basis request - 502b8cb5f0cb-gq8fmvfprog9
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found peer 'private' for 'private' from xxx.xxx.xxx.xxx:2048
> [2012-08-15 15:50:20] VERBOSE[9001] netsock2.c: == Using SIP RTP CoS mark 5
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 9
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 0
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 8
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 3
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 99
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 108
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 18
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found RTP audio format 101
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format G722 for ID 9
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format PCMU for ID 0
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format PCMA for ID 8
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format GSM for ID 3
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format G726-32 for ID 99
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format AAL2-G726-32 for ID 108
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format G729 for ID 18
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c: Found audio description format telephone-event for ID 101
> [2012-08-15 15:50:20] WARNING[9001] chan_sip.c: We are requesting SRTP for audio, but they responded without it! ;???
> [2012-08-15 15:50:20] VERBOSE[9001] chan_sip.c:
> <--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:2048 --->
> SIP/2.0 488 Not acceptable here
> And call is not accepted
> --
> Daniel
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