July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference and Speex Wideband audio quality problem
by Richard Alam 23 Nov '11

23 Nov '11

Hi Everyone, Hopefully, the answer to my problem isn't in the book as I've yet to order it. :) First off, I'm one ot the devs of BigBlueButton (http://code.google.com/p/bigbluebutton/) We currently use Asterisk for voice conference. Leif Jackson made BBB work with FreeSWITCH. Now, from BBB, a user can join a voice conference in FS. We have a java SIP client (based on Red5Phone) that calls FS. All is working well for Ulaw. I am now trying to make Speex WB work since Flash Player (where our client runs) supports Speex WB. Using Speex WB frees the java SIP client from transcoding between Nellymoser and Ulaw. The java SIP client just takes the speex audio data from RTMP and sends it to FS through RTP. However, the sound quality is bad. It is choopy and robotic. Let me describe what I have done and what I found from debugging so far. I modified vars.xml to add the speex codec. ================ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM"/> Setup the conference in dialplan/public.xml ================== <extension name="bbb_conferences"> <condition field="destination_number" expression="^(7\d{4})$"> <action application="answer"/> <action application="conference" data="$1@wideband"/> </condition> </extension> When I start up freeswitch, it looks like it's loading the speex codec fine. Full console log is here http://pastebin.com/rnHfgQYj =============== 2010-07-30 16:04:10.620375 [NOTICE] switch_loadable_module.c:251 Adding Application 'AvoidingDeadlock' 2010-07-30 16:04:10.620658 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g723_1.so **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.620726 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g729.so **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621189 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_amr.so **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621547 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_ilbc] 2010-07-30 16:04:10.621618 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2010-07-30 16:04:10.621647 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2010-07-30 16:04:10.622043 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_speex] 2010-07-30 16:04:10.622108 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2010-07-30 16:04:10.622135 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2010-07-30 16:04:10.622160 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2010-07-30 16:04:10.622513 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h26x] When I call into the conference, it looks like mod_conference does not use speex codec but L16@16000hz. ====== 2010-07-30 16:05:56.874138 [DEBUG] sofia.c:4317 Channel sofia/external/75115(a)192.168.0.166 entering state [completed][200] 2010-07-30 16:05:56.874138 [DEBUG] switch_core_session.c:647 Send signal sofia/external/75115(a)192.168.0.166 [BREAK] 2010-07-30 16:05:56.874138 [DEBUG] switch_channel.c:2494 (sofia/external/75115(a)192.168.0.166) Callstate Change RINGING -> ACTIVE 2010-07-30 16:05:56.874138 [NOTICE] mod_dptools.c:746 Channel [sofia/external/75115(a)192.168.0.166] has been answered EXECUTE sofia/external/75115(a)192.168.0.166 conference(75115@wideband) 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5237 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5282 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.877940 [DEBUG] switch_core_codec.c:116 sofia/external/75115(a)192.168.0.166 Push codec L16:10 tport_wakeup_pri(0x95bfd98): events IN Is this the correct behavior? I've tried using xlite to connect to the conference, and the audio is good. Also, when connecting from our client, the audio file that gets played by FS to notify how many users are in the conference is good. It's my voice that's bad. I tried to use the echo application instead of the conference. When I call from the BBB client, the sound quality is good. I appreciate any tips on how I can fix the problem or proceed to figure out how to fix it. Thank you very much. Richard -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton

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Help with choppy audio after attended transfer
by Chris Cureau 17 Nov '11

17 Nov '11

I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine. Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911 I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening. Any ideas?

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switch_rtp.c - Auto Changing Port
by Ross McKillop 27 Oct '11

27 Oct '11

Hi, I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic. No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console; > 2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263 After which the call works perfectly. Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start? Any assistance you can offer would be much appreciated. Best Regards, Ross

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Re: [Freeswitch-users] Different SAS with ZRTP
by Banhi Dutra 19 Oct '11

19 Oct '11

Hi all. Could anyone help me with ZRTP configuration ? For me all works fine, except for SAS exchange, which are different. I'm using default configuration because it is a test environment. Where I could find configuration or documentation about "Trusted MITM" ? Thank you. Banhi

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playing files through a remote URL
by Holger Esser 06 Oct '11

06 Oct '11

Hello, I am trying to write a simple ivr script that answers the call, plays a greeting and collects a few digits. Is it possible to fetch the audio through a remote URL (i.e. http://messageServer/en.us/greeting.wav<http://messageserver/en.us/greeting.wav> )? Or does it have to reside on the local file system or a mapped drive? Kind regards, Holger

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Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > 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 > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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Problem with Freeswitch/Sangoma D100.
by Ricardo Martinez 24 Sep '11

24 Sep '11

Hello list. I have the next problem with my Freeswitch. I’m using a Sangoma D100 transcoding card, but ‘im having problems with G729. This is my scenario. GatewayA ---------> FreeSwitch+D100 -----------> GatewayB (10.0.0.220) (10.0.0.148) (10.0.0.222) Gateway A is calling through Freeswitch to Gateway B. The problem is with the “200 - OK” message that goes from FreeSwitch to Gateway A. The “200 - OK” coming from Gateway B has the parameter “a=fmtp:18 annexb=no” in the SDP, but Freeswitch is not attaching this parameter to the “Leg A” and this is causing a one-way audio in the call. (please see the attached sdp-problem.jpg file). The weird thing is when i unload the “mod_sangoma_codec” and load the “mod_g729”, this time the “200 OK” message from the Freeswitch is using the “a=fmtp:18 annexb=no” to the Leg A, and the call is successfully established without one way audio problem. I really don’t know what could be happening, I ask the Sangoma support but thay said that this is a Freeswitch bug. Can someone help me here? These are part of my configuration files. “default.xml” <extension name="to_prueba2"> <condition field="network_addr" expression="^10\.0\.0\.220$"> <action application="set" data="call_timeout=50"/> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="nolocal:absolute_codec_string=G729,G723"/> <action application="set" data="sip_invite_domain=10.0.0.222"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:18 G729/8000,a=fmtp:18 annexb=no"/> <action application="bridge" data=" sofia/$${domain}/${destination_number}(a)10.0.0.222"/> <action application="answer"/> </condition> </extension> <extension name="to_prueba1"> <condition field="network_addr" expression="^10\.0\.0\.222$"> <action application="set" data="call_timeout=50"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=10.0.0.220"/> <action application="bridge" data=" sofia/$${domain}/${destination_number}(a)10.0.0.220"/> <action application="answer"/> </condition> </extension> Part of the “interior.xml” file:  <param name="inbound-late-negotiation" value="true"/>  <param name="inbound-codec-negotiation" value="generous"/> Y finalmente en los codec preferentes tengo (vars.xml) <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G723,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,G723"/> Thanks in advance. Regards, Ricardo.-

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how to redirect $${sounds_dir} ?
by jesse 22 Sep '11

22 Sep '11

I would like to pint sounds_dir to a different location, but it seems freeswitch doesn't support --sounds_dir as command line arguments. please let me know how to re-define sounds_dir? thanks!

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ESL Inbound - what would be the best known way to wait in XML dialplan for any command execution via ESL?
by Anton VG 19 Sep '11

19 Sep '11

Hi! Switched from outbound ESL to an INBOUND mode, for controlling the incoming CALL flow: so when call comes, Inbound ESL app catches corresponding event and executes the necessary command on the caller UUID. My question is: In XML dialplan, when call comes, if do not PARK it or do PAUSE, the ESL controlling application is too late to execute park before XML dialplan ends, and disconnect the call. PARK shows itself as not a good solution, since if call did not handled by ESL - it stales. PAUSE - i would not like to add extra delays to call handling. I suppose there should be a known good way to handle in XML dialplan if anything else executed on the given channel/uuid? Regards, Anton.

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Re: [Freeswitch-users] play_and_get_digits question
by Madovsky 17 Sep '11

17 Sep '11

sorry I think I really didn't undertand the play_and_get_digits concept. is anyone can help ? I just want to use this app to check a pin code, hangup if wrong and continue dialplan if ok Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 1:01 PM Subject: Re: play_and_get_digits question or maybe this channel var read_result is the result of play_and_get_digits ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:43 PM Subject: Re: play_and_get_digits question Ok I think I misunderstood the concept of var where digits should be put in. it's with this var I need to ckeck after this application. self forum thank you ;) ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:29 PM Subject: play_and_get_digits question I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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