- FreeSWITCH-users - FreeSWITCH Mailing Lists

Questions about speex
by Tamas 09 Feb '09

09 Feb '09

Hello, I'm looking for the best codec/scenario for the last-mile and checked speex codec capabilities (http://www.speex.org/comparison/nb_codecs.png) This will be FS-FS interconnect (where one side uses portaudio, aka a simple softphone). As I see, it would be worth to use Speex in VBR mode and also to turn on DTX. Would it be possible with current mod_speex? As I saw, currently vad and vbr options are set to 0. Would setting these hardwired options to 1 make the trick? I was thinking about G.726-32 as it was suggested last week on irc, but it has still too high bandwidth requirements and Speex seems to have better MOS values over 10kbps (or 6kbps with VBR). Thanks in advance, Tamas

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Freeswitch not processing calls
by Ankit Gandhi 09 Feb '09

09 Feb '09

Recently I noticed that fs (1.0.3 RC1) is not processing the calls. Previously about 50-60 calls were remaining active at any time. But then suddenly, only 5-6 active calls were there. So I checked sip INVITE's in ngrep, and I noticed that originator were sending CANCEL after INVITE. So I contacted them and they were telling that while sending calls, your gateways time outs after a call is sent, and so they send CANCEL after INVITE. My architecture is as follows : originator -> switch (our fs) -> terminator. CPU : Intel(R) Xeon(R) CPU X3220 @ 2.40GHz OS : Ubuntu 6.06.1 LTS 32-bit RAM : 4 GB We had also set ulimits on command prompt every time before starting fs with nc mode as follows : ulimit -c unlimited; ulimit -n 999999; ulimit -s 244; /usr/local/freeswitch/bin/freeswitch -nc In the ngrep trace, I noticed that fs was not sending INVITE to terminator in between the INVITE and CANCEL from originator. So what could be the reason for INVITE not being processed by fs and getting timeout from originator. Here is the sip trace for a sample call. Check the duration between INVITE and CANCEL from originator. -------------------------------------------------------------------- U 2009/02/07 13:16:44.354443 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 INVITE sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>. Content-Type: application/sdp. CSeq: 1 INVITE. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Contact: sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone. Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 317. . v=0. o=MG4000|2.0 492612 492612 IN IP4 66.151.208.138. s=-. c=IN IP4 66.151.208.138. t=0 0. m=audio 38832 RTP/AVP 97 18 98 96 101 13. a=rtpmap:97 G.729b/8000. a=rtpmap:98 G.729a/8000. a=rtpmap:96 G.729ab/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=fmtp:18 annexb=yes. a=ptime:20. a=rtpmap:13 CN/8000. # U 2009/02/07 13:16:44.354633 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Content-Length: 0. U 2009/02/07 13:16:59.208008 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 CANCEL sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>. CSeq: 1 CANCEL. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 0. . # U 2009/02/07 13:16:59.208095 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 200 OK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. CSeq: 1 CANCEL. Content-Length: 0. . # U 2009/02/07 13:16:59.208134 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. U 2009/02/07 13:16:59.253482 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 ACK sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>;tag=7B6p78S5SDF0j. CSeq: 1 ACK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Max-Forwards: 70. Content-Length: 0. ------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21909687.h… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Problems passing arguments to lua
by Nik Middleton 09 Feb '09

09 Feb '09

Hi Guys, I'm having some issues passing an argument to an lua script. Basically in an originate command I pass the name of a .wav file If I hard code the file session:streamFile("myfile.wav"]); It works, However, using this: session:streamFile(argv[1]); causes this error 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: No such file or directory Any Ideas? Regards

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Mod_native_file FreeSWITCH files (and script) available
by Kristian Kielhofner 09 Feb '09

09 Feb '09

Hello everyone, Now that I've got FreeSWITCH compiling under AstLinux I'm starting to look at ways to optimize FreeSWITCH. First things first: minimize transcoding. I hate transcoding. I modified a concept I came up with back in the day for Asterisk. I've created a script to convert WAV files to FreeSWITCH mod_native_file formats for various codecs. Of course I also updated the wiki: http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_f… As you can see I've also converted the current FreeSWITCH prompts and MOH into various file formats (including G.729 and G.723). These files have undergone very little testing (basically none) so I would appreciate some feedback on them. I'm also working on G.729, iLBC, speex (VBR?!?), and G.722. What about Siren, CELT, and some of the others? I don't have much experience with these and I could certainly use some help. P.S. - I'm currently using sox and Asterisk res_convert to do the conversion. I don't know how else to legitimately (and consistently) convert to these various formats. Without bringing up the usual G.729/G.723 discussions, I'm in the US and this is how I could legally do all of these conversions. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Struggling with Originate
by Nik Middleton 09 Feb '09

09 Feb '09

Hi Guys, I'm placing calls ok by using the event socket. However, I need to modify the To: Sip header prior to the call going out for routing purposes. Is it possible to do this in the Originate action? If not, can someone explain if it's possible to trigger part of the dial plan externally? I can then modify the headers and then place the call/ Regards,

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Transcoding G723
by Ali Al-Rubaie 08 Feb '09

08 Feb '09

Hi, I need FreeSWITCH to transcode from G711 to G723 but I couldn't do so because it supports G723 only in passthru mode. So, what's the solution? Thanks,

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Seeking opinion on shared disk space
by Woody Dickson 08 Feb '09

08 Feb '09

Hi, In my deployment scenario, I plan to have two redundant freeswitch servers running on two different boxes. Two key features I am leveraging on freeswitch are voicemail and call recording and playback., and as a result of that, a shared storage for playback of the recorded wav files is needed. When the user traffic is high, I am affraid that NFS or even GFS can't scale well. On the other hand, a real SAN hardware with optical-fabric is too expensive for us. We are therefore considering using iSCSI SAN to build a cheap SAN for that purpose. Does anyone have experience setting up a shared storage between multiple freeswitch servers and can share some inputs with me? Thanks for all your help. Woody

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Re: [Freeswitch-users] mod_shout delay in trunk
by freeswitch-users＠digitaldan.com 07 Feb '09

07 Feb '09

That worked great! I wanted to say just how awesome Freeswitch is, I have been doing voip related development with SIP since 2000 and this is by far the most well designed piece of voip software I have used or developed on. I currently have a homegrown sip server built on the NIST sip stack with Sun's JMF libraries for RTP processing. 95% of the code and complexity is handling the SIP and RTP sessions, the other 5% is the final application logic and what is most important to me . By letting freeswitch do whats its good at (call routing, sip and media handling) it allows me to focus on what I'm good at (what should we do with those streams, like record them). I have been bragging about this project to anybody who will listen! Dan- ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, < freeswitch-users(a)digitaldan.com > wrote: For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" < brian(a)freeswitch.org > To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users(a)digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ 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 -- 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 _______________________________________________ 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

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need suggestion on using mod_easyroute
by Juan Backson 07 Feb '09

07 Feb '09

Hi, I notice that there is a newly-developed mod_easyroute model available. Has anyone used it with large amount of routes ( ex > 1M ) on a high traffic scenario? For that kind of scenario, would it be better to consider using out-going event socket to serve that purpose? I would greatly appreciate any recommendation. Thanks, JB

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Call accounting - CDR's
by Nik Middleton 07 Feb '09

07 Feb '09

Hi Guys I'm looking for some pointers on how to collect CDR's and store in mysql. Is there anything built in yet? I can rate the calls as a batch process, I simply need the call data. Regards

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