[asterisk-dev] SIP Secure and Annouced Transfert Asterisk 1.8Trunk.
Bernard Merindol(F)
Bernard.Merindol at free.fr
Mon Feb 28 02:10:37 CST 2011
Hi all,
I have installed the last 1.8 branch with correction on SIP REFER and dead lock.
The following issue has been RESOLVED.
======================================================================
https://issues.asterisk.org/view.php?id=18468
This patch resolve announced transfer in SIP mode with directmedia=yes.
Its good and thank for this job at all.
BUT this patch not resolv my problem on announced transfert with SIPS and SRTP.
I got the same error after finish transfert:
[Feb 28 08:39:57] WARNING[5735] res_srtp.c: SRTP unprotect: authentication failure
[Feb 28 08:39:57] WARNING[5735] res_srtp.c: SRTP unprotect: authentication failure
[Feb 28 08:39:57] WARNING[5735] res_srtp.c: SRTP unprotect: authentication failure
[Feb 28 08:39:57] WARNING[5735] res_srtp.c: SRTP unprotect: authentication failure
[Feb 28 08:39:57] WARNING[5735] res_srtp.c: SRTP unprotect: authentication failure
[Feb 28 08:39:57] WARNING[5735] res_srtp.c: SRTP unprotect: authentication failure
And the audio is one way.
My new question is:
Is it possible with SRTP to use announced transfert, the key of SRTP follow the transfert or not ?
It's a BUG or it's not possible ? If it's a BUG I open a report.
Thank for your helps.
Best regards
Bernard
On 23 févr. 2011, at 23:18, Alec Davis wrote:
> Yes the patch bug18837.diff3.txt at
> https://issues.asterisk.org/view.php?id=18837 does address the deadlock
> issue.
>
> I responded to your initial email as you posted the deadlock file
> transferts-bug-lock.txt on https://issues.asterisk.org/view.php?id=18468
> which has the same deadlock as in bug report 18837.
>
> Others: I'm now happy with the revised patch 'diff3', Please test and
> comment, and hopefully approve.
>
> 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: Wednesday, 23 February 2011 6:04 a.m.
>> To: Asterisk Developers Mailing List
>> Subject: Re: [asterisk-dev] SIP Secure and Annouced Transfert
>> Asterisk 1.8Trunk.
>>
>> 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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