Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
Thanks I will have a deeper look and asking concrete questions afterwords. Thanks! MatzeMuc86 Von: devel-boun...@lists.opensips.org [mailto:devel-boun...@lists.opensips.org] Im Auftrag von Dave Singer Gesendet: Freitag, 25. Februar 2011 18:26 An: OpenSIPS devel mailling list Cc: MatzeMuc86 Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server MatzeMuc86, Opensips handles just the SIP signaling which contains the information about where the RTP should connect to. The modules media_proxy and nat_helper can be used to communicate to the external applications media_proxy and rtp_proxy respectively that setup proxying of the RTP and return to the opensips module the connection setup it has prepared for the RTP. The module then alters the SDP appropriately and the opensips script continues on deciding where to send and sending the SIP message to the next server. I believe rtp_proxy can be setup to stream audio from a file and it might be a starting point for you to mix the stereo audio. However I'm not sure if opensips can be the endpoint of a call (SIP) without writing or extending a module for that purpose. The B2B module is something you might look at to see how much tweaking it would take to make it do what you want. Dave On Fri, Feb 25, 2011 at 1:15 AM, MatzeMuc86 matzemu...@gmail.com wrote: Hi, as I already invested a lot of time, I try to be sure to check out all possibilities. I already know about PJSIP but that does not mean that OpenSIPS could not be the right project for me - I wanted to be sure. Anyway: Thanks A LOT for your very nice support and searching time!!! -Ursprüngliche Nachricht- Von: devel-boun...@lists.opensips.org [mailto:devel-boun...@lists.opensips.org] Im Auftrag von Saúl Ibarra Corretgé Gesendet: Freitag, 25. Februar 2011 10:08 An: OpenSIPS devel mailling list Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server Hi, On 02/25/2011 10:02 AM, MatzeMuc86 wrote: Hello Adrian, I know that I need some RTP part which receives, mixes and sends the media stream. I thought I can do all these things with OpenSIPS? I saw that SDP is implemented, but, of course, this is transported by SIP. Maybe I am wrong about RTP and OpenSIPS - sorry. SDP is just signaling, OpenSIPS doesn't deal with RTP at all. Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. - but no stereo. TO implement this, the project is that big that - after talking with the freeswitch developers - this seems to be a very big project. As it is only a Bachelor Thesis I thought about finding an easier way to implement my idea. IIRC, PJSIP does have stereo support to some degree. A quick search returned this: http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/ Regards, -- Saúl Ibarra Corretgé AG Projects ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
try to check http://www.iptel.org/sems On 25.02.2011 12:02, MatzeMuc86 wrote: Hello Adrian, I know that I need some RTP part which receives, mixes and sends the media stream. I thought I can do all these things with OpenSIPS? I saw that SDP is implemented, but, of course, this is transported by SIP. Maybe I am wrong about RTP and OpenSIPS - sorry. Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. - but no stereo. TO implement this, the project is that big that - after talking with the freeswitch developers - this seems to be a very big project. As it is only a Bachelor Thesis I thought about finding an easier way to implement my idea. Thanks MatzeMuc86 -Ursprüngliche Nachricht- Von: Adrian Georgescu [mailto:a...@ag-projects.com] Gesendet: Donnerstag, 24. Februar 2011 23:42 An: matzemu...@gmail.com; OpenSIPS devel mailling list Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server OpenSIPS is a SIP server, it handles the SIP signaling, not the media. To implement what you describe you need something that can create and handle incoming media streams and is able to mix them. If FreeSwitch or Asterisk cannot already do these things close to what you plan to implement, you will need to reinvent some wheels. Adrian On Feb 24, 2011, at 10:43 PM, MatzeMuc86 wrote: Hello, I try to develop a stereo back to back user agent conference server where simultaneously not only mono but also stereo clients can connect. B2BUA to be as much standard conform as possible (it's just an idea). Stereo clients means that the client sends mono to the server but gets a stereo signal back (upmixed by some tricks; see HRTF). As I already tried to develop this idea at freeswitch which seems not be be possible, I want to know from some opensips expters if this is possible with the effort of a Bachelor Thesis (I want to program my own, of course). -- С уважением, директор ООО Еврокомм Гамов Виктор attachment: vit.vcf___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
Thanks I will have a deeper look and asking concrete questions afterword. Thanks! MatzeMuc86 Von: devel-boun...@lists.opensips.org [mailto:devel-boun...@lists.opensips.org] Im Auftrag von Dave Singer Gesendet: Freitag, 25. Februar 2011 18:26 An: OpenSIPS devel mailling list Cc: MatzeMuc86 Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server MatzeMuc86, Opensips handles just the SIP signaling which contains the information about where the RTP should connect to. The modules media_proxy and nat_helper can be used to communicate to the external applications media_proxy and rtp_proxy respectively that setup proxying of the RTP and return to the opensips module the connection setup it has prepared for the RTP. The module then alters the SDP appropriately and the opensips script continues on deciding where to send and sending the SIP message to the next server. I believe rtp_proxy can be setup to stream audio from a file and it might be a starting point for you to mix the stereo audio. However I'm not sure if opensips can be the endpoint of a call (SIP) without writing or extending a module for that purpose. The B2B module is something you might look at to see how much tweaking it would take to make it do what you want. Dave On Fri, Feb 25, 2011 at 1:15 AM, MatzeMuc86 matzemu...@gmail.com wrote: Hi, as I already invested a lot of time, I try to be sure to check out all possibilities. I already know about PJSIP but that does not mean that OpenSIPS could not be the right project for me - I wanted to be sure. Anyway: Thanks A LOT for your very nice support and searching time!!! -Ursprüngliche Nachricht- Von: devel-boun...@lists.opensips.org [mailto:devel-boun...@lists.opensips.org] Im Auftrag von Saúl Ibarra Corretgé Gesendet: Freitag, 25. Februar 2011 10:08 An: OpenSIPS devel mailling list Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server Hi, On 02/25/2011 10:02 AM, MatzeMuc86 wrote: Hello Adrian, I know that I need some RTP part which receives, mixes and sends the media stream. I thought I can do all these things with OpenSIPS? I saw that SDP is implemented, but, of course, this is transported by SIP. Maybe I am wrong about RTP and OpenSIPS - sorry. SDP is just signaling, OpenSIPS doesn't deal with RTP at all. Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. - but no stereo. TO implement this, the project is that big that - after talking with the freeswitch developers - this seems to be a very big project. As it is only a Bachelor Thesis I thought about finding an easier way to implement my idea. IIRC, PJSIP does have stereo support to some degree. A quick search returned this: http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/ Regards, -- Saúl Ibarra Corretgé AG Projects ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
Hi, On 02/25/2011 10:02 AM, MatzeMuc86 wrote: Hello Adrian, I know that I need some RTP part which receives, mixes and sends the media stream. I thought I can do all these things with OpenSIPS? I saw that SDP is implemented, but, of course, this is transported by SIP. Maybe I am wrong about RTP and OpenSIPS - sorry. SDP is just signaling, OpenSIPS doesn't deal with RTP at all. Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. - but no stereo. TO implement this, the project is that big that - after talking with the freeswitch developers - this seems to be a very big project. As it is only a Bachelor Thesis I thought about finding an easier way to implement my idea. IIRC, PJSIP does have stereo support to some degree. A quick search returned this: http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/ Regards, -- Saúl Ibarra Corretgé AG Projects ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
Hi, as I already invested a lot of time, I try to be sure to check out all possibilities. I already know about PJSIP but that does not mean that OpenSIPS could not be the right project for me - I wanted to be sure. Anyway: Thanks A LOT for your very nice support and searching time!!! -Ursprüngliche Nachricht- Von: devel-boun...@lists.opensips.org [mailto:devel-boun...@lists.opensips.org] Im Auftrag von Saúl Ibarra Corretgé Gesendet: Freitag, 25. Februar 2011 10:08 An: OpenSIPS devel mailling list Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server Hi, On 02/25/2011 10:02 AM, MatzeMuc86 wrote: Hello Adrian, I know that I need some RTP part which receives, mixes and sends the media stream. I thought I can do all these things with OpenSIPS? I saw that SDP is implemented, but, of course, this is transported by SIP. Maybe I am wrong about RTP and OpenSIPS - sorry. SDP is just signaling, OpenSIPS doesn't deal with RTP at all. Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. - but no stereo. TO implement this, the project is that big that - after talking with the freeswitch developers - this seems to be a very big project. As it is only a Bachelor Thesis I thought about finding an easier way to implement my idea. IIRC, PJSIP does have stereo support to some degree. A quick search returned this: http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/ Regards, -- Saúl Ibarra Corretgé AG Projects ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
MatzeMuc86, Opensips handles just the SIP signaling which contains the information about where the RTP should connect to. The modules media_proxy and nat_helper can be used to communicate to the external applications media_proxy and rtp_proxy respectively that setup proxying of the RTP and return to the opensips module the connection setup it has prepared for the RTP. The module then alters the SDP appropriately and the opensips script continues on deciding where to send and sending the SIP message to the next server. I believe rtp_proxy can be setup to stream audio from a file and it might be a starting point for you to mix the stereo audio. However I'm not sure if opensips can be the endpoint of a call (SIP) without writing or extending a module for that purpose. The B2B module is something you might look at to see how much tweaking it would take to make it do what you want. Dave On Fri, Feb 25, 2011 at 1:15 AM, MatzeMuc86 matzemu...@gmail.com wrote: Hi, as I already invested a lot of time, I try to be sure to check out all possibilities. I already know about PJSIP but that does not mean that OpenSIPS could not be the right project for me - I wanted to be sure. Anyway: Thanks A LOT for your very nice support and searching time!!! -Ursprüngliche Nachricht- Von: devel-boun...@lists.opensips.org [mailto:devel-boun...@lists.opensips.org] Im Auftrag von Saúl Ibarra Corretgé Gesendet: Freitag, 25. Februar 2011 10:08 An: OpenSIPS devel mailling list Betreff: Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server Hi, On 02/25/2011 10:02 AM, MatzeMuc86 wrote: Hello Adrian, I know that I need some RTP part which receives, mixes and sends the media stream. I thought I can do all these things with OpenSIPS? I saw that SDP is implemented, but, of course, this is transported by SIP. Maybe I am wrong about RTP and OpenSIPS - sorry. SDP is just signaling, OpenSIPS doesn't deal with RTP at all. Refering to FreeSWITCH: IT can handle RTP, conferences, B2BUA eg. - but no stereo. TO implement this, the project is that big that - after talking with the freeswitch developers - this seems to be a very big project. As it is only a Bachelor Thesis I thought about finding an easier way to implement my idea. IIRC, PJSIP does have stereo support to some degree. A quick search returned this: http://blog.pjsip.org/2008/03/31/doing-it-in-stereo/ Regards, -- Saúl Ibarra Corretgé AG Projects ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] Sip stereo B2BUA conference server
Hello, I try to develop a stereo back to back user agent conference server where simultaneously not only mono but also stereo clients can connect. B2BUA to be as much standard conform as possible (it's just an idea). Stereo clients means that the client sends mono to the server but gets a stereo signal back (upmixed by some tricks; see HRTF). As I already tried to develop this idea at freeswitch which seems not be be possible, I want to know from some opensips expters if this is possible with the effort of a Bachelor Thesis (I want to program my own, of course). Thansk a lot MatzeMuc86 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Sip stereo B2BUA conference server
OpenSIPS is a SIP server, it handles the SIP signaling, not the media. To implement what you describe you need something that can create and handle incoming media streams and is able to mix them. If FreeSwitch or Asterisk cannot already do these things close to what you plan to implement, you will need to reinvent some wheels. Adrian On Feb 24, 2011, at 10:43 PM, MatzeMuc86 wrote: Hello, I try to develop a stereo back to back user agent conference server where simultaneously not only mono but also stereo clients can connect. B2BUA to be as much standard conform as possible (it's just an idea). Stereo clients means that the client sends mono to the server but gets a stereo signal back (upmixed by some tricks; see HRTF). As I already tried to develop this idea at freeswitch which seems not be be possible, I want to know from some opensips expters if this is possible with the effort of a Bachelor Thesis (I want to program my own, of course). Thansk a lot MatzeMuc86 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
