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

Tue Feb 22 11:04:18 CST 2011

Good Morning,

your patch is on deadlock issue ?

In SIPS with branch version don't have deadlock but have no media from C to A.

I think is an encryption problem . 

This problem is very blocking for me, is not possible to deploy my new project.

Best regards
Bernard

On 21 févr. 2011, at 19:12, Alec Davis wrote:

> Bernard: Please check bug https://issues.asterisk.org/view.php?id=18837
> There you will find bug18837.diff2.txt
> 
> Others: Please review the patch and comment. It's tested and works, but I
> don't actually like the unlock and relock of the 'pvt' that I've done around
> the call to 'transmit_reinvite_with_sdp'().
> 
> It's a simple deadlock between 
> === Thread ID: -1292625008 (do_monitor started at [24470] chan_sip.c
> restart_monitor())
> === ---> Lock #0 (chan_sip.c): MUTEX 23964 handle_request_do &netlock
> 0xb6796e80 (1)
> === ---> Lock #1 (channel.c): MUTEX 6211 ast_do_masquerade channels
> 0x8d4e0c8 (1)
> === ---> Lock #2 (channel.c): MUTEX 6214 ast_do_masquerade original
> 0xbd98f48 (1)
> === ---> Lock #3 (channel.c): MUTEX 6234 ast_do_masquerade clonechan
> 0xb24bf7d0 (1)
> === ---> Waiting for Lock #4 (chan_sip.c): MUTEX 6164 sip_fixup p 0xb24bab10
> (1)
> === --- ---> Locked Here: chan_sip.c line 27632 (sip_set_rtp_peer)
> 
> === -------------------------------------------------------------------
> ===
> === Thread ID: -1315861616 (pbx_thread started at [ 5035] pbx.c
> ast_pbx_start())
> === ---> Lock #0 (chan_sip.c): MUTEX 27632 sip_set_rtp_peer p 0xb24bab10 (1)
> === ---> Waiting for Lock #1 (pbx.c): MUTEX 9467 pbx_builtin_getvar_helper
> chan 0xb24bf7d0 (1)
> === --- ---> Locked Here: channel.c line 6234 (ast_do_masquerade)
> 
> Alec Davis
> 
>> -----Original Message-----
>> From: asterisk-dev-bounces at lists.digium.com 
>> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of 
>> Bernard Merindol(F)
>> Sent: Tuesday, 22 February 2011 6:09 a.m.
>> To: Asterisk Mailing List Developers
>> Subject: [asterisk-dev] SIP Secure and Annouced Transfert 
>> Asterisk 1.8 Trunk.
>> 
>> 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
>> 
>> 
>> 
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> 
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