July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch
by Stanislav Sinyagin 01 Jul '14

01 Jul '14

Kees, do I understand it correctly that you're only sending audio towards the SIP endpoint, so it's not a bidirectional conversation? In this case, you can utilize audio streaming via HTTP, probably it's the easiest way to deliver audio toward FreeSWITCH: session.streamFile("shout://some.server.com/file.mp3", "") https://wiki.freeswitch.org/wiki/Mod_shout ________________________________ From: Kees Jan Koster <kjkoster(a)java-monitor.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>; ssinyagin(a)yahoo.com Sent: Tuesday, July 1, 2014 1:27 PM Subject: Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch Dear Stanislav, I was rather hoping to avoid that path. My C is not all that great. :-/ But if I have to... Any endpoints that come close that I might want to look at? Kees Jan > AFAIK the RTP parameters and streaming are handled from the endpoint module: > https://wiki.freeswitch.org/wiki/Modules > > so, you might end up in writing your own module for what you want. > > > From: Kees Jan Koster <kjkoster(a)kjkoster.org> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Tuesday, July 1, 2014 9:21 AM > Subject: [Freeswitch-users] handling signalling externally, RTP via freeswitch > > Dear All, > > I need some reading advice/pointers on the following: I want to use FreeSwitch as a bridge from a custom client to SIP. The SIP part is easy and works. The non-SIP part is the challenge. > > I have my audio sending client hooked up via mod_event_socket. I really like the way events and ESL work in FreeSwitch, by the way. Good control and easy to code. But I digress. So I have my client hooked up over ESL for signalling and need to know where to send the RTP stream to from the non-SIP client side. > > Call flow (A-leg is the custom client, B-leg is sofia-SIP) > [1] custom client starts call to SIP side over ESL [1] > FreeSwitch handles the signalling with the SIP client > SIP client answers the call > [2] FreeSwitch sends event which tells the client where the RTP stream should go to > custom client starts sending RTP audio > FreeSwitch bridges the two legs and people talk > either side hangs up and the call is done > > The reverse (SIP dialling the custom client) won't happen and does not have to be supported. > > My concrete question is how [1] should happen. The custom client should issue an "bgapi originate". The arguments for the SIP side are clear, but what to use for the custom client? What should I fill in for [1], below? > > bgapi originate [1] sofia/internal/1002(a)example.com > > -- > Kees Jan > > http://java-monitor.com/ > kjkoster(a)kjkoster.org > +31651838192 > > The secret of success lies in the stability of the goal. -- Benjamin Disraeli > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 -- Kees Jan kjkoster(a)java-monitor.com http://java-monitor.com/ +31651838192 The secret of success lies in the stability of the goal. -- Benjamin Disraeli

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handling signalling externally, RTP via freeswitch
by Kees Jan Koster 01 Jul '14

01 Jul '14

Dear All, I need some reading advice/pointers on the following: I want to use FreeSwitch as a bridge from a custom client to SIP. The SIP part is easy and works. The non-SIP part is the challenge. I have my audio sending client hooked up via mod_event_socket. I really like the way events and ESL work in FreeSwitch, by the way. Good control and easy to code. But I digress. So I have my client hooked up over ESL for signalling and need to know where to send the RTP stream to from the non-SIP client side. Call flow (A-leg is the custom client, B-leg is sofia-SIP) [1] custom client starts call to SIP side over ESL [1] FreeSwitch handles the signalling with the SIP client SIP client answers the call [2] FreeSwitch sends event which tells the client where the RTP stream should go to custom client starts sending RTP audio FreeSwitch bridges the two legs and people talk either side hangs up and the call is done The reverse (SIP dialling the custom client) won't happen and does not have to be supported. My concrete question is how [1] should happen. The custom client should issue an "bgapi originate". The arguments for the SIP side are clear, but what to use for the custom client? What should I fill in for [1], below? bgapi originate [1] sofia/internal/1002(a)example.com -- Kees Jan http://java-monitor.com/ kjkoster(a)kjkoster.org +31651838192 The secret of success lies in the stability of the goal. -- Benjamin Disraeli

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Re: [Freeswitch-users] Recording in flac format ?
by julien terrasson 01 Jul '14

01 Jul '14

Hi Brian, I'm still interested in recording in flac format but i'm not very familiar with mod_sndfile.c (and not familiar with c programming either). To get started, i tried to introduce the flac extension the same way oga extension was introduced : } else if (!strcmp(ext, "oga")) { context->sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS; context->sfinfo.samplerate = handle->samplerate; } else if (!strcmp(ext, "flac")) { context->sfinfo.format = SF_FORMAT_FLAC | SF_FORMAT_PCM_16; context->sfinfo.samplerate = handle->samplerate; } And i also added the extension in the list : char *extras[] = { "r8", "r16", "r24", "r32", "gsm", "ul", "ulaw", "al", "alaw", "adpcm", "vox","flac", NULL }; then make & make install. But my first attempt was not really successfull : 2014-07-01 11:47:36.670463 [ERR] mod_sndfile.c:205 Error Opening File [/var/callid:84/record0.flac] [File contains data in an unimplemented format.] Does anybody have docs/guidelines on how to use libsndfile ? Regards, J.terrasson ..Adding formats would need you to modify mod_sndfile to enable them. --- Brian West From: julien terrasson <julien.terrasson(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Cc: Date: Wed, 18 Jun 2014 14:58:22 +0200 Subject: [Freeswitch-users] Fwd: Recording in flac format ? Hi, I'm using freeswitch version 1.4.5 I would like to record a session in .flac format. I assume this could be somehow possible since FS rely on libsndfile-1.0.26pre5 and libsndfile claims flac support since v1.0.18. I tried to compiled FS with instruction from description of https://jira.freeswitch.org/browse/FS-1197 : Apt-get install libflac-dev and "./configure --enable-flac".. However that didn't help, flac extension is still not usable on FS recording app. Does anyone have an idea how to have flac format enabled ? Regards, J.Terrasson

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RTCP and OPUS
by Kamil Nigmatullin 01 Jul '14

01 Jul '14

Hello all. What I basicly want is to have real adaptive codec on both sides (client and FS), that could exchange link quality information through RTCP. As far as I see, after a brief looking through some parts of source code of FS, mod_opus works as static codec and don't change its behavior when Internet link is changing because it has no communication with RTCP module. Could anybody of developers recommend me the most efficient way to deploy this communication process through Freeswitch Core? Is there a way to export and import information from/to module/core during the call? Thanks in advance.

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Adding new Gateway with registration false - incoming calls works but outgoing calls not
by K K 01 Jul '14

01 Jul '14

Hello all, I have the following problem. I have added a new Gateway (SIP Trunk) on my FS simply by adding new configuration file in the "external" folder, new dialplan file in the "dialplan\default" folder and adding the CIDR of my Gateway as the allowed domain in the acl.conf.xml. And it works perfectly. Then I have change my configuration - I have added the "register false" parameter into my Gateway configuration and the result is that I am not longer able to call any of my Gateway SIP endpoint from FS, but FS is still able to receive the incoming calls from the Gateway, so I have lost the outgoing calls. I wondering if such configuration is in fact supported? I have 2 book about FS and in both of them there are only examples with registration set to true. I was looking in google for some help, but I have only found that I need to set the call authentication to false - but this was my default setting actually. Thank you for help in that topic.

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