- FreeSWITCH-users - FreeSWITCH Mailing Lists

Interop list for crypto-enabled devices
by Alois Komenda 01 Feb '08

01 Feb '08

Hello, is there a document listing all devices (softphones and hardphones) which work with FreeSWITCH in encrypted (TLS + SRTP) mode? If not, could you please tell me with which devices you tested encrypted calls and what the results of these test are? Who of you got eyeBeam working with FS when encrypting signalling or/and media? Thanks a lot -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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Re: [Freeswitch-users] XML CDRs
by Anthony Minessale 30 Jan '08

30 Jan '08

The callflows are stacked with most recent first. created_time, answered_time and hangup_time are only set once and copied to every subsequent callflow so if you want, you can just look at the top callflow tag from there the in the <times> you can look at created answered hangup etc.... Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, January 29, 2008 3:34:49 PM Subject: [Freeswitch-users] XML CDRs Hi - Having installed mod_limit, my XML CDRs now sometimes - if the limit's exceeded - have two <callflow> sections, and I guess there can be an arbitrary number of them if a call's transferred from extension to extension, or whatever. My problem is knowing how to chain them together in the right order. The <times> section in each contains a <transfer_time> which I can use with the <profile_created_time> to chain them together - at least in the examples I've got - but this is a bit fiddly. Am I missing something obvious? --Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 8692 0677 M: 07773 800623 http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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XML CDRs
by David Knell 30 Jan '08

30 Jan '08

Hi - Having installed mod_limit, my XML CDRs now sometimes - if the limit's exceeded - have two <callflow> sections, and I guess there can be an arbitrary number of them if a call's transferred from extension to extension, or whatever. My problem is knowing how to chain them together in the right order. The <times> section in each contains a <transfer_time> which I can use with the <profile_created_time> to chain them together - at least in the examples I've got - but this is a bit fiddly. Am I missing something obvious? --Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 8692 0677 M: 07773 800623 http://www.3c.co.uk

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I need help to handle incoming calls from Cisco AS5300
by Steve Xu 29 Jan '08

29 Jan '08

Hi Folks, I have a Cisco AS5300 in a different domain as FS, It will send calls to FS with format ( prefix + phone number ) through sip with codec G723. here is the configuration setting on FS. Set the default sip_profile without auth ( no need register) <param name="accept-blind-reg" value="true"/> Set the default dial-plan with expression to handle prefix <condition field="destination_number" expression="^9090180012345678$"> <action application="bridge" data=" sofia/default/180012345678(a)sip.gateway.com"/> </condition> </extension> what else i need to setup? AS5300 keep getting error code 27. Thanks for any help, StevenLee

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Audio rate changes and clipping on playback
by freeswitch＠dalethatcher.com 28 Jan '08

28 Jan '08

Hello all! I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host. My test setup is: Client <-> NAT ADSL <-> Freeswitch I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk. I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok. So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format: $ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others). Is anyone else seeing the same issues or has a solution? thanks, - Dale

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Re: [Freeswitch-users] Audio rate changes and clipping on playback
by Anthony Minessale 28 Jan '08

28 Jan '08

yes please install our ssh key and send me the details. the key is in the svn tree in ssh_keys/shinzon.pub you can find us on irc all day or im or email me. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: "freeswitch(a)dalethatcher.com" <freeswitch(a)dalethatcher.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, January 28, 2008 8:01:00 AM Subject: Re: [Freeswitch-users] Audio rate changes and clipping on playback I mean it sounds like parts of the sound are missing. e.g. the frequency of playback seems to change and I can hear a warbling/slow down from the Freeswitch instance on the audio samples. If you (or any others on the list) have a spare couple of minutes then I'm happy to send you the details of my test server. It would be reassuring if someone else could hear and verify the issue. thanks, - Dale On Mon, Jan 28, 2008 at 02:05:20PM +0100, Sluschny, Thomas wrote: > don't know what you mean with clipping, but I've notice clipping errors when I play sound files with maximum levels with G.711, clipping means that on max. posivite value a max. negative value is played. > this is true only on G.711, e. g. Linear16 codes are fine. > > regards, > thomas > > > > -----Ursprüngliche Nachricht----- > Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von freeswitch(a)dalethatcher.com > Gesendet: Samstag, 26. Januar 2008 16:30 > An: freeswitch-users(a)lists.freeswitch.org > Betreff: [Freeswitch-users] Audio rate changes and clipping on playback > > Hello all! > > I'm experiencing some problems with clipping and the rate of playback of > wav files from Freeswitch. I've not raised this before as I thought the > problems were due to hosting over an ADSL line. However I've now seen > the same issues running from a dedicated host. > > My test setup is: > > Client <-> NAT ADSL <-> Freeswitch > > I've replicated using both the Gizmo and FWD clients from Windows. If I > substitute Asterisk with Freeswitch then the audio comes over fine. The > two test samples are one of myself and the fpm-sunshine.wav that is > bundled with asterisk. > > I've tried playing around with the codecs but see the same problems > regardless. Also I've performed a test with a conference hosted on the > Freeswitch instance with myself and friend dialling in using Gizmo and > the audio is fine. Which implied to me that the link between Gizmo to > Freeswitch is ok. > > So I'm guessing that there is something wrong with the samples that is > causing the problem? They're both in this format: > > $ file fpm-sunshine.wav > RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz > > My deployment platform is Debian stable and the Freeswitch SVN revision > is 7343. I've tried looking in Jira but keep getting a system error > page (for search terms 'audio', 'wav' and others). > > Is anyone else seeing the same issues or has a solution? > > thanks, > > - Dale > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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MIKEY-Support
by Alois Komenda 28 Jan '08

28 Jan '08

Hello, does FreeSWITCH support MIKEY? If not, is it scheduled? Which key exchange methods does it support? What about the soft and hard phones on the market? What's the preferred method for key exchange there? Regards, -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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get numbers for dialthrough
by Dominik 'friibii' Friberg 28 Jan '08

28 Jan '08

hi all, i'm newbee on freeswitch, just moved from asterisk to it.. i like to rewrite my "call-trhough" - but i don't have a clue how to "ask for the number to dial".. any help? thanks and happy weekend friibii

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Re: [Freeswitch-users] Call Waiting not working!!!!!
by Anthony Minessale 25 Jan '08

25 Jan '08

Your issue has been fixed in the latest release. Also, Can you please avoid cross-posting all of your inquiries to both the users and dev list. It's time consuming to go through the same emails twice. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Irshad Ahamed <irshad.fswitch(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Friday, January 25, 2008 6:17:58 AM Subject: Re: [Freeswitch-users] Call Waiting not working!!!!! I have just verified. Call Waiting works fine with openzap @ rev 361 but it is not working with the latest openzap trunk @ rev 379. -Irshad. On Jan 25, 2008 4:00 PM, Irshad Ahamed < irshad.fswitch(a)gmail.com> wrote: Hi, I am working on Freeswitch@ rev 7309. I remember with earlier revision call waiting was working fine. But now on getting the second incoming call FXS is returning 502 (BAD gateway error). Was there any change on FXS side pertaining to number of calls allowed or something??? Freeswitch Log: 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(dialed_ext=6601) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [dialed_ext]=[6601] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(hangup_after_bridge=true) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [hangup_after_bridge]=[true] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(bypass_media=false) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [bypass_media]=[false] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute bridge(openzap/1/1) 2008-01-25 15:48:17 [ERR] mod_openzap.c:833 channel_outgoing_channel() No channels available 2008-01-25 15:48:17 [ERR] switch_ivr_originate.c:827 switch_ivr_originate() Cannot Create Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_ivr_originate.c:1226 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [INFO] mod_dptools.c:1325 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [NOTICE] mod_dptools.c:1352 audio_bridge_function() Hangup sofia/default/6502@192.168.2.51:5060 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/6502@192.168.2.51:5060 [KILL] 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/6502@192.168.2.51:5060 [BREAK] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/6502@192.168.2.51:5060 Running State Change CS_HANGUP 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/6502@192.168.2.51:5060) State HANGUP 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/6502@192.168.2.51:5060 hanging up, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:332 sofia_on_hangup() Responding to INVITE with: 502 2008-01-25 15:48:17 [DEBUG] sofia.c:183 sofia_event_callback() event [nua_i_state] status [502][Call state] session: sofia/default/6502@192.168.2.51:5060 2008-01-25 15:48:17 [DEBUG] sofia.c:1430 sofia_handle_sip_i_state() Channel sofia/default/6502@192.168.2.51:5060 entering state [terminated] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/6502@192.168.2.51:5060, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Locked, Waiting on external entities 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Ended 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/6502@192.168.2.51:5060 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:665 process_event() EVENT [ONHOOK][1:1] STATE [UP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:691 process_event() Changing state on 1:1 from UP to DOWN 2008-01-25 15:48:36 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:1037 on_fxs_signal() got FXS sig [STOP] 2008-01-25 15:48:36 [NOTICE] mod_openzap.c:1115 on_fxs_signal() Hangup OpenZAP/777 [CS_EXECUTE] [NORMAL_CLEARING] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill OpenZAP/777 [KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:649 zap_analog_channel_run() ANALOG CHANNEL thread ended. 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:186 audio_bridge_thread() read: OpenZAP/777 Bad Frame.... Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/777] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:693 switch_ivr_multi_threaded_bridge() sofia/default/777(a)sipx.oneconv.com State Change CS_LOOPBACK -> CS_RESET 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() OpenZAP/777 Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/777) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:418 channel_on_hangup() OpenZAP/777 CHANNEL HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP OpenZAP/777, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 1 (OpenZAP/777) Locked, Waiting on external entities 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:180 audio_bridge_thread() write: OpenZAP/777 Bad Frame....[160] Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/777(a)sipx.oneconv.com] 2008-01-25 15:48:36 [NOTICE] switch_ivr_bridge.c:257 audio_bridge_on_loopback() Hangup sofia/default/777(a)sipx.oneconv.com [CS_RESET] [NORMAL_CLEARING] 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 1 (OpenZAP/777) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel OpenZAP/777 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/777(a)sipx.oneconv.com [KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/777(a)sipx.oneconv.com Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/777(a)sipx.oneconv.com) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/777(a)sipx.oneconv.com hanging up, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:325 sofia_on_hangup() Sending BYE to sofia/default/777(a)sipx.oneconv.com 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/777(a)sipx.oneconv.com , cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com ) Locked, Waiting on external entities 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/777(a)sipx.oneconv.com [CS_HANGUP] Thanks in advance... BR, Irshad. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-users] MIKEY-Support
by Anthony Minessale 25 Jan '08

25 Jan '08

The hard part was getting the srtp seamlessly integrated into our RTP stack. We had 3 brands of phones to test with while implementing and this was was the only way they all supported. There is clearly a feud on how to exchange keys out there and we will end up implementing each of them as they become popular. As a developer, I can only implement what I have to test with. Do you know any devices that support MIKEY to test against? There is a LGPL libmikey out there so it's a strong possibility we can implement it as long as we have a test bed and some indication that it will be widely accepted and desired by people. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Alois Komenda <alois.komenda(a)esk.fraunhofer.de> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, January 25, 2008 6:20:19 AM Subject: Re: [Freeswitch-users] MIKEY-Support How can you ever be sure TLS is really used end-to-end? Even if TLS is used "end-to-end" i.e. on every hop, every involved proxy can read your keys. So if you can trust all proxys that route your messages, SDES is secure. -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK -----Ursprüngliche Nachricht----- Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Brian West Gesendet: Freitag, 25. Januar 2008 12:09 An: freeswitch-users(a)lists.freeswitch.org Betreff: Re: [Freeswitch-users] MIKEY-Support How on earth is it not secure? The keys are exchanged over a secure TLS channel. That is secure. Read section 8.3 again. "Thus, IT IS REQUIRED that MIME secure multiparts, IPsec, TLS, or some other data security service be used to provide message authentication for the encapsulating protocol that carries the SDP messages having a crypto attribute (a=crypto)." It does however say in 8.3 "When the communication path of the SDP message is routed through intermediate systems that inspect parts of the SDP message, security protocols such as [IPsec] or TLS SHOULD NOT be used for encrypting and/ or authenticating the security description." This can clearly be seen don't trust it if TLS isn't used end to end for the sip signaling channel. SDES seems to be the most widely used method at this point as you pointed out. I feel the security afforded by using SDES + TLS is way more than you'll ever get elsewhere. We do accept patches. ;) /b On Jan 25, 2008, at 12:15 AM, Alois Komenda wrote: > I don't think SDES over TLS can be called secure. And according to RFC > 4568 this combination should not be used. > (Anyway this seems to be the mostly used configuration at the moment.) > > Even if MIKEY is not a perfect soloution for the problem, it provides > end-to-end security for keying material. > > Regards > > -- > Alois Komenda > Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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