- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Portaudio AEC take 2
by João Mesquita 15 Apr '13

15 Apr '13

Guys, I've been looking at an AEC solution to integrate on portaudio for the softphones to take advantage of it and it seems like WebRTC has one that could be used. I am not by any means an expert on the matter so before I start hacking things, I would like to ask your opinion. First of all, licensing... Is this license compatible? http://www.webrtc.org/license-rights/license If the license is compatible, where would be the best place to implement it? I would definitely need some pointers.. I found out that Blink is using it on PJMEDIA as this patch ( http://sipsimpleclient.org/projects/sipsimpleclient/repository/entry/patche…) shows. PJMedia already has an interface for noise supression and AEC so I guess it was "easy" to implement there. We would implement it as a media bug or on the portaudio module itself? Also, if anyone knows if this will bring good results, would be great! AEC that just works is something that is really wanted on a softphone... João Mesquita

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Sound Not Ok(Choppy Sound) after 250 ports
by Subhash 12 Apr '13

12 Apr '13

Hi, I ran the load test, raising the port count slowly (every 5 to 10 minutes or so) to 100, 150, 200, 250, 275 and 300 ports. There were no audio problems until I reached 275 ports. After probably 10 minutes at 275 ports, audio problems started, and after that they were consistent. Of course I can’t guarantee that there were no audio problems at lower port counts - if the problem only occurred intermittently I might not have caught it or I might not have waited long enough. Once the audio problems started, I slowly lowered the number of ports back to 250 ports. The audio problems disappeared after about 2 1/2 minutes at 250 ports. I then increased the number of ports back to 275 ports. Thanks, Subhash.

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FreeSWITCH Conference Call Today
by Michael Collins 10 Apr '13

10 Apr '13

Hello all! Come join the FreeSWITCH conference call at 1PM EDT/10AM PDT: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10 We have a few items to discuss, and we'll be asking the community for input on an important question: How do you choose whether to do closed, freemium, or open source when you've built a great solution? Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Weekly News and Notes
by Michael Collins 08 Apr '13

08 Apr '13

A belated happy Monday to all! Last week on our conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_03>we spent some time discussing various topics such as Jira system improvements and updates, sound prompts, ClueCon 2013 and several other topics. The recordings can be found in the usual place<http://torrents.freeswitch.org/conf_call_2013-04-03.torrent>. On this week's call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10>we are going to have a community discussion and a few minutes of Q&A from the audience. If possible, please add your questions to the agenda page and we'll research them prior to the call. Where applicable we'll ask members of the audience to update the wiki to reflect any undocumented knowledge that has been discussed. Things have been busy with the advent of ClueCon 2013<http://www.cluecon.com>season but we're on top of things and we'll keep everyone posted on all the particulars. Feel free to register at any time. Also, please contact us at this email address if you have any questions about being a speaker, sponsor, or attendee. We'll be glad to assist. One last item: I wanted to personally say thank you to Steven Ayre for all of his hard work with answering questions on the mailing list. He has done a great job of helping lots of people with a variety of questions. Many thanks to Steven and all the others who make FreeSWITCH such a great FOSS community. Thanks and have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Development Info
by Georgi Stefanov 04 Apr '13

04 Apr '13

Thanks Steven, I am looking the DOCs too, Currently I succeeded to place my non-sip call into the dialplan. I see the dialplan execution is started but when playback execution is started is stops with hanging up my call. ( I have tried to pre-answer but...) I need something like " your_io_routines.outgoing_channel is called create a new session (with switch_core_session_request()) parse outbound_profile->destination_number to get called number qualify the session, attached channel and attached private data The channel is currently in CS_NEW Change channel state to CS_INIT Make the phone ring (and call switch_channel_mark_ring_ready()) your_state_handler.on_init is called Change channel state to CS_ROUTING your_state_handler.on_routing is called What to do here? The call is answered switch_channel_mark_answered(channel) " the info from Authoring_Freeswitch_Modules but with more details In my case: I have no calling of your_io_routines.outgoing_channel I do the codec staff in on_init function On_Routing I do the pre-answer On_Receiving I try to start rtp session I can not understand am I doing it right or not ? (probably not, but I need to do it the right way, but I do not know the right way) So every kind of info is welcome ----------------------------------------------------------------- Намери автомобил от най-бързоразвиващия се сайт за коли в България! http://www.carmarket.bg/

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Compilation fails on FreeBSD
by Adrian Gschwend 03 Apr '13

03 Apr '13

Hi group, git master does not compile on FreeBSD right now, did a clean checkout followed by bootstrap & configure: ... quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -Ilibs/sofia-sip/libsofia-sip-ua/sdp -Ilibs/sofia-sip/libsofia-sip-ua/su -I/usr/local/include -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/libtpl-1.5/src -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/local/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -I/usr/local/include -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_rtp.lo -MD -MP -MF .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_rtp.o cc1: warnings being treated as errors /usr/local/src/freeswitch/libs/srtp/crypto/include/datatypes.h:412: warning: 'be64_to_cpu' defined but not used gmake[1]: *** [libfreeswitch thanks Adrian

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Come join the conference call!
by Michael Collins 03 Apr '13

03 Apr '13

We have several items to discuss, so please join us! http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_03 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Development Info
by Georgi Stefanov 02 Apr '13

02 Apr '13

Hello All, It is spring time (almost) again and as most of the things I need to reproduce or simply develop other kind of endpoint. The purpose: 1.I need other kind of signaling, a protocol which is not supported by FS 2.The need is the call to enter into the dialplan and execute it (FS to execute the dialplan a call from other signalization point is received) What I have done till now: I started from http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Before_You_Beg…... I have used mod_sofia and mod_skypopen as reference as result I am (the call is) into the dialplan now (using switch_core_session_execute_exten) but I can not deal properly with channel state callbacks and no RTP data is flowing/traveling/sending. I have been given some hints to check-out rtp.c but now I think I need more info about creating channels and dealing with their states. Then I will need to deal with the RTP data So if anyone have some clues and wants to share...I will be very glad

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ClueCon 2013 - Call For Speakers
by Michael Collins 02 Apr '13

02 Apr '13

[image: https://mail-attachment.googleusercontent.com/attachment/u/0/?ui=2&ik=43fec…]<http://www.cluecon.com> ClueCon - the open source IP communications conference by developers, fordevelopers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea fora technical presentation for ClueCon 2013 then we would like to hear about it. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon *loves *technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools, especially those in a production environment - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations - We are especially interested in WebRTC-related talks and demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include the following items: - Working title - Brief description of the talk (abstract) - Name of the presenter Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time. ClueCon 2013 Registration Information ClueCon 2013 registration is now open!. Visit the registration page<http://www.cluecon.com/register?cc12cfs> for details. Be sure to book your room at the Hyatt Chicago Magnificent Mile<http://www.cluecon.com/hotels/>and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Sending RTP data without SIP
by Georgi Stefanov 28 Mar '13

28 Mar '13

Hello All, Recently, I am having fun with freeswitch. I need to implement a module, which sends RTP data to previously specified IP and PORT. I do not want to have SIP/SDP session. Which is the best or just good enough way to do that? I understand my module should act as an endpoint then I need probably would need to create the session object, set the needed codecs, load (for example) the desired file for playing and start sending RTP. I had some debugging actions on freeswitch and found that functions sofia_glue_negotiate_sdp, sofia_glue_tech_choose_port are dealing with local and remote site of RTP connections. Could you advice me ?

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