[ https://issues.asterisk.org/jira/browse/ASTERISK-20234?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Matt Jordan updated ASTERISK-20234:
-----------------------------------
Assignee: tootai
Status: Waiting for Feedback (was: Triage)
> 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
> Assignee: 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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