[asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8 Trunk.

[Bernard Merindol(F)]

Hi All,

I continue to test Asterisk 1.8 for announced Transfert.

I works with the Trunk version 

Connected to Asterisk SVN-trunk-r308371 currently running on c3devsecure

For normal SIP I have a work around for announced transfert, if I configure all phones without directmedia (directmedia=no) the announced transfert works fine. With direct media not works see 

https://issues.asterisk.org/view.php?id=18468

But, if I test the same configuration with SIPS and SRTP the announced transfert not works 
The tree phones is configured with
encryption=yes
directmedia=no
transport=tls

A Call B
B Annouce to C
When B finish Transfert, the channels is connected between A to C but the RTP (SRTP in this case) is not works or works only beetwen A to C. Newer audio form C to A.

On full we see :

[Feb 21 17:54:47] DEBUG[15231] chan_sip.c: Sip transfer:--------------------
[Feb 21 17:54:47] DEBUG[15231] chan_sip.c: -- Transferer to PBX channel: SIP/1001-0000004b State Up
[Feb 21 17:54:47] DEBUG[15231] chan_sip.c: -- Transferer to PBX second channel (target): SIP/1001-0000004c State Up
[Feb 21 17:54:47] DEBUG[15231] chan_sip.c: -- Bridged call to transferee: SIP/1000-0000004a State Up
[Feb 21 17:54:47] DEBUG[15231] chan_sip.c: -- Bridged call to transfer target: SIP/1002-0000004d State Up
[Feb 21 17:54:47] DEBUG[15231] chan_sip.c: -- END Sip transfer:--------------------


[Feb 21 17:54:47] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:47] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure

[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: SIP response 200 to RE-invite on outgoing call 474496ed441d4f0636c4e0c410f10ffe at 192.168.169.60:5061
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 192.168.169.211... UNSUPPORTED.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED.
[Feb 21 17:54:47] DEBUG[15227] netsock2.c: Splitting '192.168.169.211' gives...
[Feb 21 17:54:47] DEBUG[15227] netsock2.c: ...host '192.168.169.211' and port '(null)'.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.169.211... OK.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 21 17:54:47] VERBOSE[15227] chan_sip.c: [Feb 21 17:54:47] Found RTP audio format 8
[Feb 21 17:54:47] DEBUG[15227] rtp_engine.c: Setting payload 8 based on m type on 0xb50b8fdc
[Feb 21 17:54:47] VERBOSE[15227] chan_sip.c: [Feb 21 17:54:47] Found RTP audio format 101
[Feb 21 17:54:47] DEBUG[15227] rtp_engine.c: Setting payload 101 based on m type on 0xb50b8fdc
[Feb 21 17:54:47] VERBOSE[15227] chan_sip.c: [Feb 21 17:54:47] Found audio description format PCMA for ID 8
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 21 17:54:47] VERBOSE[15227] chan_sip.c: [Feb 21 17:54:47] Found audio description format telephone-event for ID 101
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 21 17:54:47] DEBUG[15227] res_srtp.c: Policy already exists, not re-adding
[Feb 21 17:54:47] WARNING[15227] sip/sdp_crypto.c: Could not set local SRTP policy
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SkxXKDBrRzh1YzchbUZnKTk8a1RKUmEjfDNNUWAo... UNSUPPORTED.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 21 17:54:47] DEBUG[15227] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.


[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure
[Feb 21 17:54:59] WARNING[15703] res_srtp.c: SRTP unprotect: authentication failure


 I search to get the old version with asterisk 1.6 to tes but the svn not works

svn co http://svn.digium.com/svn/asterisk/team/group/srtp_reboot/ asterisk-srtp
svn: URL 'http://svn.digium.com/svn/asterisk/team/group/srtp_reboot' doesn't exist

Thank for your help.

Best regards
Bernard Merindol