[Freeswitch-users] Bridging to a non SIP based system
Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging to a non SIP based system
you could make an endpoint module for FS that speaks the special protocol then use that to call the conference. On Fri, Dec 4, 2009 at 3:29 PM, Phillip Jones pjinthe...@gmail.com wrote: Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging to a non SIP based system
I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging to a non SIP based system
Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo m...@giagnocavo.netwrote: I think you will need to sort out the signaling first, as you’ll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip Jones *Sent:* Friday, December 04, 2009 2:29 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging to a non SIP based system
Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you're going to end up playing with pointers except the syntax will be more verbose). Then again, I have a natural aversion to C so maybe it's just me ;) -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 3:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Bridging to a non SIP based system Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo m...@giagnocavo.netmailto:m...@giagnocavo.net wrote: I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bridging to a non SIP based system
You can re-use some of mod_sofia's functions (like sofia_glue_parse_sdp) and only write the part of signalling thats different from SIP. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 4-Dec-09, at 8:52 PM, Michael Giagnocavo wrote: Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you’re going to end up playing with pointers except the syntax will be more verbose). Then again, I have a natural aversion to C so maybe it’s just me ;) -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 3:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Bridging to a non SIP based system Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo m...@giagnocavo.net wrote: I think you will need to sort out the signaling first, as you’ll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
