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Query related to enabling SRTP in FreeSWITCH-1.0.7
by Goutham BG 22 Dec '10

22 Dec '10

Posting the below query to freeswitch-users list as well. Any hints will be really helpful. ---------- Forwarded message ---------- From: Goutham BG <bggoutham(a)gmail.com> Date: Mon, Dec 20, 2010 at 9:16 PM Subject: Query related to enabling SRTP in FreeSWITCH-1.0.7 To: freeswitch-dev(a)lists.freeswitch.org Hi, I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing some issues. I have the following entry in my dialplan XML file: *<extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="bridge" data="loopback/app=socket: 47.152.232.156:8084 async full"/> </condition </extension>* A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. But the media is established in SRTP in one way and RTP in the other way. The phone offers the following SDP in the INVITE message: v=0 o=- 10170 10170 IN IP4 47.152.232.147 s=Sip Call c=IN IP4 47.152.232.147 t=0 0 m=audio 5016 RTP/AVP 0 8 18 101 102 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 X-nt-inforeq/8000 a=sendrecv m=audio 5016 RTP/SAVP 0 8 18 101 102 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 X-nt-inforeq/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+ +ornJoXjZ5tSadho4 a=sendrecv As we can see, there are two "m=" lines in the SDP of the offer; one for RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP: v=0 o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156 s=FreeSWITCH c=IN IP4 47.152.232.156 t=0 0 m=audio 11280 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=audio 0 RTP/SAVP 19 As you can see above, FreeSWITCH accepts the RTP stream and rejects the SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send the media in RTP since it accepted RTP in the answer (200OK). *Query: ======* In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK), I made the following change(i.e, setting sip_secure_media=true) in FS dial plan: *<extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="loopback/app=socket: 47.152.232.156:8084 async full"/> </condition </extension>* In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown below: m=audio 0 RTP/AVP 19 m=audio 12084 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned dial plan change (i.e, setting sip_secure_media=true) is not working. It is still behaving in the same way as it did without the XML change. Can you please let me know if anything else needs to be added in dialplan XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing something here? I have referred the following FS wiki pages for making the SRTP changes: http://wiki.freeswitch.org/wiki/Secure_RTP http://wiki.freeswitch.org/wiki/SRTP Note: There is no issue when the SIP phone is configured in "SRTP only" mode where only SRTP stream is offered in the SDP of the INVITE. In this case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't require setting "sip_secure_media=true" in the dialplan XML file. P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking basic questions. Thanks Goutham B G

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2600hz. Access to my web interface.
by ZILOT 22 Dec '10

22 Dec '10

Hello !! I have installed FS+GUI (2600hz) on FreeBSD http://www.2600hz.org/ (http://www.2600hz.org/) I have forgotten the password to my web interface (2600hz), but i have ssh root access. Where I can search login\password to my web interface or replace them? Please help.

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SBC on kamailio
by Henry Dogger 22 Dec '10

22 Dec '10

Hi all, We have a kamailio installation for our voip system and would like to add a SBC. We found the freeswitch possibility: http://wiki.freeswitch.org/wiki/SBC_Setup But in this case, 302 Redirect SIP is used, we would like the SBC to act as a pass-through controller, just to be a gateway for SIP clients from outside. Is something like this possible? And will registrations be handled correctly since all SIP messages will be on port 5060. Thanks in advance. Kind regards, Henry Dogger Telecats BV

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XMAS Greets
by Helmut Kuper 22 Dec '10

22 Dec '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello dear FS Community, because today is my last day at work for this year (hurray) I would like to wish you all a peacefull christmas and a healthy new year 2011. Thanks a lot for all the tips&tricks, bugfixes and new features this year. Greets from verrrrrry cold and snowy germany Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk0SKXwACgkQ4tZeNddg3dy+BACeMt0qUo+s6Qq6KKT9zYI5dKys 7nkAn0coBygql4jAPCAfm60B6FfGC7/U =UboS -----END PGP SIGNATURE-----

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How to select and play audio in a conference
by Fabio Bigliardi 22 Dec '10

22 Dec '10

Hi all, I would like to select audio files to play to all members of a conference from within the conference through the use of DTMF keypad inputs. How can I do this? Thank you very much. Fabio Bigliardi :: SYSNET TELEMATICA srl :: CONFIDENZIALE: Questo messaggio e gli eventuali allegati sono confidenziali e riservati. Se vi è stato recapitato per errore e non siete fra i destinatari elencati, siete pregati di darne immediatamente avviso al mittente e cancellare il messaggio di posta e gli eventuali file allegati. Le informazioni contenute non devono essere mostrate ad altri, né utilizzate, memorizzate o copiate in qualsiasi forma. CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and delete this e-mail and any attachment. Do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium.

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Fosdem 2011
by Michal Bielicki 22 Dec '10

22 Dec '10

Hi everybody, I will be running a freeswich booth at fosdem 2011 in Brussels, Belgium (http://www.fosdem.org/) If anybody is interested to meet, participate etc, just drop me an email. I will see if we can create some flyers for the show and a nice poster for the back of the booth. Ideas, helping hands, human booth decoration or any other help are greatly appreciated. See you in Brussels Michal Bielicki Geschäftsführer / CEO Seventh Signal Ltd. & Co. KG Weigandufer 45, Büro 115 D-12059 Berlin Voice: +49 30 60988730 Amtsgericht Berlin Charlottenburg HRA 44413 B, Ust.-ID: DE266981999 Geschäftsführer: Michal Bielicki Persönlich Haftende Gesellschafterin: Seventh Signal Ltd, 69 Great Hampton St. Birmingham, B18 6EW, GB, Company Nr.: 06889439 WWW.: http://www.seventhsignal.de

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Support for RTCP report via SIP PUBLISH
by Helmut Kuper 22 Dec '10

22 Dec '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I want to collect RTCP reports from snom devices via FS's event system. Unfortunately it doesn't support it. Maybe because it is still a draft: http://tools.ietf.org/html/draft-ietf-sipping-rtcp-summary-05 I wonder whether this feature is planed in near future. Nevertheless I hacked sofia.c and sofia-presence.c to get it working just for testing. Here is the "git diff": - ---DIFF------------------------------------------------------------------------ diff --git a/src/mod/endpoints/mod_sofia/sofia.c b/src/mod/endpoints/mod_sofia/sofia.c index 2e13b54..3622506 100644 - --- a/src/mod/endpoints/mod_sofia/sofia.c +++ b/src/mod/endpoints/mod_sofia/sofia.c @@ -1530,6 +1530,7 @@ void *SWITCH_THREAD_FUNC sofia_profile_thread_run(switch_thread_t *thread, void TAG_IF(profile->pres_type, NUTAG_ALLOW_EVENTS("include-session-description")), TAG_IF(profile->pres_type, NUTAG_ALLOW_EVENTS("presence.winfo")), TAG_IF(profile->pres_type, NUTAG_ALLOW_EVENTS("message-summary")), + TAG_IF(profile->pres_type, NUTAG_ALLOW_EVENTS("vq-rtcpxr")), NUTAG_ALLOW_EVENTS("refer"), SIPTAG_SUPPORTED_STR(supported), SIPTAG_USER_AGENT_STR(profile->user_agent), TAG_END( switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Set params for %s\n", profile->name); diff --git a/src/mod/endpoints/mod_sofia/sofia_presence.c b/src/mod/endpoints/mod_sofia/sofia_presence.c index 143666c..c998280 100644 - --- a/src/mod/endpoints/mod_sofia/sofia_presence.c +++ b/src/mod/endpoints/mod_sofia/sofia_presence.c @@ -2679,7 +2679,17 @@ void sofia_presence_handle_sip_i_publish(nua_t *nua, sofia_profile_t *profile, n pd_dup = strdup(payload->pl_data); - - if ((xml = switch_xml_parse_str(pd_dup, strlen(pd_dup)))) { + //EWETEL RTCP hack + //event_type = sip_header_as_string(profile->home, (void *) sip->sip_event); + if (strcmp(sip_header_as_string(profile->home, (void *) sip->sip_event), "vq-rtcpxr")==0) { + if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) { + switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "event_type", "vq-rtcpxr"); + switch_event_add_body(event, "%s", pd_dup); + switch_event_fire(&event); + } + } + //DONE + else if ((xml = switch_xml_parse_str(pd_dup, strlen(pd_dup)))) { char *open_closed = "", *note_txt = ""; if (sip->sip_user_agent) { - ---DIFF END----------------------------------------------------------- It simply allows SIP PUBLISH events of type "vq-rtcpxr" and adds the sip payload to the internal event body. The corresponding event is this: Content-Length: 1178 Content-Type: text/event-plain Event-Name: PRESENCE_IN Core-UUID: f852daae-6da9-4979-8dc8-fa11651a7891 FreeSWITCH-Hostname: test FreeSWITCH-IPv4: 1.2.3.4 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-12-21%2015%3A19%3A09 Event-Date-GMT: Tue,%2021%20Dec%202010%2014%3A19%3A09%20GMT Event-Date-Timestamp: 1292941149981062 Event-Calling-File: sofia_presence.c Event-Calling-Function: sofia_presence_handle_sip_i_publish Event-Calling-Line-Number: 2685 event_type: vq-rtcpxr Content-Length: 703 VQSessionReport LocalMetrics: Timestamps:START=2010-12-21T14:19:08Z STOP=2010-12-21T14:19:10Z SessionDesc:PT=9 PD=G.722 PPS=50 SSUP=off CallID:3c2f99c5dc8f-k1jiuwkd70m9 x-UserAgent:snom370/8.4.22 FromID:"Helmut Kuper" <sip:8000@sipdomain.de> ToID:<sip:***6@sipdomain.de;user=phone> LocalAddr:IP=85.16.245.234 PORT=10568 SSRC=0xF919B973 RemoteAddr:IP=85.16.246.16 PORT=19770 SSRC=0x DialogID:3c2f99c5dc8f-k1jiuwkd70m9;to-tag=7HU6y795c8r8B;from-tag=rbxbm5jjme x-SIPmetrics:SVA=RG SRD=890 SFC=0 x-SIPterm:SDC=OK SDR=OR JitterBuffer:JBA=0 JBR=0 JBN=0 JBM=0 JBX=65535 PacketLoss:NLR=0.0 JDR=0.0 BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=2440 GMIN=16 Delay:RTD=0 ESD=0 IAJ=0 QualityEst:MOSLQ=4.2 MOSCQ=4.1 regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk0QuwgACgkQ4tZeNddg3dx6/gCgij7KZDD0qHLFkujDjzH/hYu+ 7UEAn2SM1eFQsPA6ICNxtexEprdWyGCk =kbkn -----END PGP SIGNATURE-----

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compile error on libmp3lame and mod_shout
by Norman Lam 22 Dec '10

22 Dec '10

Hi there I am using AMD athlon X2, windows xp sp3, VC++2008 express (only installed this software) using the Git source downloaded today the first time I am getting these 2 error , is there any ways to resolve this or work around. Please advise Norman Lam 1st error : : 47>------ Build started: Project: libmp3lame, Configuration: Debug Win32 ------ 47>Compiling... 44>s2_srvr.c 47>bitstream.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\bitstream.c': No such file or directory 47>version.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\version.c': No such file or directory 47>VbrTag.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\VbrTag.c': No such file or directory 47>vbrquantize.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\vbrquantize.c': No such file or directory 47>util.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\util.c': No such file or directory 47>takehiro.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\takehiro.c': No such file or directory 47>set_get.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\set_get.c': No such file or directory 47>reservoir.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\reservoir.c': No such file or directory 47>quantize_pvt.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\quantize_pvt.c': No such file or directory 47>quantize.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\quantize.c': No such file or directory 47>psymodel.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\psymodel.c': No such file or directory 47>presets.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\presets.c': No such file or directory 47>newmdct.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\newmdct.c': No such file or directory 47>mpglib_interface.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\mpglib_interface.c': No such file or directory 47>lame.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\lame.c': No such file or directory 47>id3tag.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\id3tag.c': No such file or directory 47>gain_analysis.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\gain_analysis.c': No such file or directory 47>fft.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\fft.c': No such file or directory 47>encoder.c 47>c1 : fatal error C1083: Cannot open source file: '..\..\lame-3.97\libmp3lame\encoder.c': No such file or directory 47>Generating Code... 47>Build log was saved at "file://c:\FS_GIT\libs\win32\libmp3lame\Debug\BuildLog.htm" 47>libmp3lame - 19 error(s), 0 warning(s) 44>s2_pkt.c : : 2nd error : : 65>.\mod_shout.c(38) : fatal error C1083: Cannot open include file: 'lame.h': No such file or directory 65>Build log was saved at "file://c:\FS_GIT\src\mod\formats\mod_shout\Win32\Debug\BuildLog.htm" 65>mod_shout - 1 error(s), 0 warning(s) : : : 140>Linking... 140>Embedding manifest... 140>Build log was saved at "file://c:\FS_GIT\w32\Console\Debug\BuildLog FreeSwitchConsole.htm" 140>FreeSwitchConsole - 0 error(s), 1 warning(s) ========== Build: 124 succeeded, 2 failed, 0 up-to-date, 14 skipped ==========

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Nokia N900 and poor Voicemail IVR audio
by Rajil Saraswat 21 Dec '10

21 Dec '10

Hi all, For some strange reason my Nokia N900 doesn't play the audio for the digits of the IVR. For example it will play "For advanced options Press" but will not play number 5. The IVR audio works fine if i call using ekiga. Is there explanation for this? Initially i thought this was my FS server issue but i have now tested it against somebody elses FS with the same response. Thanks

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forcing codec on leg b
by Sameer Khan 21 Dec '10

21 Dec '10

Hello guyz, Need your help I want my fs to behave like this when call arrives for a number 1234656789 at fs i call a php file in which i check in the database that what should be the forwarding address of this number. for example the forwarding address is 123456789(a)1.1.1.1 then i return that forwarding address in bridge application in the end i want to define some setting in sip profiles, user directory or wherever so when ever ip is 1.1.1.1 it set the codec on leg b to pcma like it is defined in asterisk as [1.1.1.1] host=1.1.1.1 type=peer disallow=all allow=alaw i want my sip profile to check if network address is 1.1.1.1, it uses a particular codec lets say pcma but if network address is 2.2.2.2 it uses the defult and this i want to define for around 40 ip addresses

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