Thread: [OpenSIPStack] OSSPhone / Solegy Dialer Registration from tag
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From: Thomas R. <tho...@it...> - 2008-08-11 13:30:31
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Hi. I've been testing solegy desktop dialer / oss phone with our SIP server and noticed that the registration fails because the REGISTER with the authorization header does have a different from-tag which we don't allow. Is there any way to override this when using just the active control? What happens exactly is the following: REGISTER from phone with From-tag 1 TRYING from Service (this is a default answer which we send immediately) 401 Unauthorized from Service with www-authenticate header REGISTER from phone with From-tag 2 TRYING from Service 401 Unauthorized Call id's are the same but from tags vary. As far as I know this check was implemented for security reasons to make sure it is the same client. Any help appreciated. Regards, Thomas Raschbacher P.S.: if i have to change this in the OSS code and re-compile the activeX control I can live with it too of course but I'd need to know what I have to do. Mit freundlichen Grüßen Thomas Raschbacher ____________________________________________ itCampus Technology GmbH Österreich * Deutschland * Italien Dresdner Straße 45 /DG 1200 Wien tho...@it... Tel: +43 (1) 890 22 82 - 58 Fax: +43 (1) 890 22 82 - 958 http://www.itctec.com UID: ATU 6339 0618 Firmenbuchnr: FN292598t, Handelsgericht Wien Geschäftsführer: Andreas Günser, Andreas Lassmann Joint Venture von itCampus und MEC itCampus Gruppe Deutschland * Großbritannien * Italien * Österreich * Schweiz * Slowakei http://www.itcampus.eu |
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From: <jo...@op...> - 2008-08-11 14:02:30
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Thomas, This was already fixed by Ilian in CVS since last May 2008. * Revision 1.69 2008/05/05 03:32:13 ijpinzon * Do not change From tag when sending a REGISTER as a response to a challenge. I just made some new changes to make 401 responses not change the from tag as well. Joegen Thomas Raschbacher wrote: > Hi. > > I've been testing solegy desktop dialer / oss phone with our SIP server and noticed that the registration fails because the REGISTER with the authorization header does have a different from-tag which we don't allow. Is there any way to override this when using just the active control? > > What happens exactly is the following: > > REGISTER from phone with From-tag 1 > TRYING from Service (this is a default answer which we send immediately) > 401 Unauthorized from Service with www-authenticate header REGISTER from phone with From-tag 2 TRYING from Service > 401 Unauthorized > > Call id's are the same but from tags vary. > > As far as I know this check was implemented for security reasons to make sure it is the same client. > > Any help appreciated. > > Regards, > Thomas Raschbacher > P.S.: if i have to change this in the OSS code and re-compile the activeX control I can live with it too of course but I'd need to know what I have to do. > > Mit freundlichen Grüßen > Thomas Raschbacher > ____________________________________________ > itCampus Technology GmbH > Österreich * Deutschland * Italien > Dresdner Straße 45 /DG > 1200 Wien > tho...@it... > Tel: +43 (1) 890 22 82 - 58 > Fax: +43 (1) 890 22 82 - 958 > http://www.itctec.com > UID: ATU 6339 0618 > Firmenbuchnr: FN292598t, Handelsgericht Wien > Geschäftsführer: Andreas Günser, Andreas Lassmann > Joint Venture von itCampus und MEC > > itCampus Gruppe > Deutschland * Großbritannien * Italien * Österreich * Schweiz * Slowakei > http://www.itcampus.eu > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > 6‹â®ë ¢éÝŠ{aŠÈ§r‰¢ž ž²Æ x(^rGo m§ÿðïÊ&Uê슉üÓ]üýX«ºÀÚµ¦Ú±í»Ó?×7Eé^jǃj×¼ÿ]?ÛM<ë]ÏM=== |
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From: Thomas R. <tho...@it...> - 2008-08-11 14:04:48
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Joegen, Thanks for your reply. Is there a new build of ossphone / solegy dialer with this fix already applied or do I have to build it myself? (If I have to build it myself, I assume I need to use CVS as 1.1.7 has been out for a while right?) Regards > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > jo...@op... > Sent: Monday, August 11, 2008 16:02 > To: ope...@li... > Subject: Re: [OpenSIPStack] OSSPhone / Solegy Dialer Registration from > tag > > Thomas, > > This was already fixed by Ilian in CVS since last May 2008. > > * Revision 1.69 2008/05/05 03:32:13 ijpinzon > * Do not change From tag when sending a REGISTER as a response to a > challenge. > > I just made some new changes to make 401 responses not change the from > tag as well. > > Joegen > > Thomas Raschbacher wrote: > > Hi. > > > > I've been testing solegy desktop dialer / oss phone with our SIP > server and noticed that the registration fails because the REGISTER > with the authorization header does have a different from-tag which we > don't allow. Is there any way to override this when using just the > active control? > > > > What happens exactly is the following: > > > > REGISTER from phone with From-tag 1 > > TRYING from Service (this is a default answer which we send > > immediately) > > 401 Unauthorized from Service with www-authenticate header REGISTER > > from phone with From-tag 2 TRYING from Service > > 401 Unauthorized > > > > Call id's are the same but from tags vary. > > > > As far as I know this check was implemented for security reasons to > make sure it is the same client. > > > > Any help appreciated. > > > > Regards, > > Thomas Raschbacher > > P.S.: if i have to change this in the OSS code and re-compile the > activeX control I can live with it too of course but I'd need to know > what I have to do. > > > > Mit freundlichen Grüßen > > Thomas Raschbacher > > ____________________________________________ > > itCampus Technology GmbH > > Österreich * Deutschland * Italien > > Dresdner Straße 45 /DG > > 1200 Wien > > tho...@it... > > Tel: +43 (1) 890 22 82 - 58 > > Fax: +43 (1) 890 22 82 - 958 > > http://www.itctec.com > > UID: ATU 6339 0618 > > Firmenbuchnr: FN292598t, Handelsgericht Wien > > Geschäftsführer: Andreas Günser, Andreas Lassmann Joint Venture > von > > itCampus und MEC > > > > itCampus Gruppe > > Deutschland * Großbritannien * Italien * Österreich * Schweiz * > > Slowakei http://www.itcampus.eu > > > > > > --------------------------------------------------------------------- > - > > --- This SF.Net email is sponsored by the Moblin Your Move > Developer's > > challenge Build the coolest Linux based applications with Moblin SDK > & > > win great prizes Grand prize is a trip for two to an Open Source > event > > anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > 6‹â®ë¢éÝŠ{aŠÈ§r‰¢ž ž²Æ x(^rGo > m§ÿðïÊ&Uê슉üÓ]üýX«ºÀÚµ¦Ú±í»Ó?×7Eé^jǃj×¼ÿ]?ÛM<ë]ÏM=== > > |
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From: Thomas R. <tho...@it...> - 2008-08-12 11:18:10
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Ok got this to work now. One other question.. I've seen the SF::SoftPhoneInterface::DoBlindTransfer method, but I'm missing methods to put calls on Hold and/or Retrieve them again? Is this currently implemented? (if not in SF::SoftPhoneInterface is it implemented in atlsip?) REgards > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > jo...@op... > Sent: Monday, August 11, 2008 16:38 > To: ope...@li... > Subject: Re: [OpenSIPStack] OSSPhone / Solegy Dialer Registration from > tag > > There is no available binary yet. You will have to grab VS and compile > the latest from CVS. > > Thomas Raschbacher wrote: > > Joegen, > > > > Thanks for your reply. Is there a new build of ossphone / solegy > dialer with this fix already applied or do I have to build it myself? > (If I have to build it myself, I assume I need to use CVS as 1.1.7 has > been out for a while right?) > > > > Regards > > > > > > > ----------------------------------------------------------------------- > -- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the > world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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From: <jo...@op...> - 2008-08-12 13:29:40
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I don't exactly remember well if this is implemented in ATLSIP or not but I am pretty sure that the underlying subsystem has support for it. However, call hold/unhold is only significant in attended call transfer. Thomas Raschbacher wrote: > Ok got this to work now. > One other question.. I've seen the SF::SoftPhoneInterface::DoBlindTransfer method, but I'm missing methods to put calls on Hold and/or Retrieve them again? Is this currently implemented? (if not in SF::SoftPhoneInterface is it implemented in atlsip?) > > REgards > > >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> jo...@op... >> Sent: Monday, August 11, 2008 16:38 >> To: ope...@li... >> Subject: Re: [OpenSIPStack] OSSPhone / Solegy Dialer Registration from >> tag >> >> There is no available binary yet. You will have to grab VS and compile >> the latest from CVS. >> >> Thomas Raschbacher wrote: >> >>> Joegen, >>> >>> Thanks for your reply. Is there a new build of ossphone / solegy >>> >> dialer with this fix already applied or do I have to build it myself? >> (If I have to build it myself, I assume I need to use CVS as 1.1.7 has >> been out for a while right?) >> >>> Regards >>> >>> >>> >> ----------------------------------------------------------------------- >> -- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > 6‹â®ë ¢éÝŠ{aŠÈ§r‰¢ž ž²Æ x(^rGo m§ÿðïÊ&Uê슉üÓ]üýX«ºÀÚµ¦Ú±í»Ó×9Eé^jǃj×¼ÿ]ÛM<ãŸOM=== |
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From: <jo...@op...> - 2008-08-11 14:37:49
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There is no available binary yet. You will have to grab VS and compile the latest from CVS. Thomas Raschbacher wrote: > Joegen, > > Thanks for your reply. Is there a new build of ossphone / solegy dialer with this fix already applied or do I have to build it myself? (If I have to build it myself, I assume I need to use CVS as 1.1.7 has been out for a while right?) > > Regards > > |
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From: Manoj K. J. <ma...@as...> - 2008-08-12 11:24:19
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Hello, I am looking forward to integrate GIPS media processing to opensipstack. Initially i want to incorporate it only on its Softphone interface and build ATLSip with it. I think i would need to make changes in SDP, Audio devices handling, Start/stop RTP and encryption. I tried to find some documentation on openSBC architecture but did not find much. Please give me some of your valuable directions on how should i start with this. Regards, Manoj |
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From: <jo...@op...> - 2008-08-12 13:26:40
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The first thing you need to do is to implement your codec as a subclass of OpalFramedTranscoder. You can check out how Speex is implemented (speexcodec.h, speexcodec.cxx) and base you custom codec from there. The next step is to call your codec registration macro in allcodecs.h. Manoj Kumar Joshi wrote: > Hello, > > I am looking forward to integrate GIPS media processing to opensipstack. > Initially i want to incorporate it only on its Softphone interface and build > ATLSip with it. > > I think i would need to make changes in SDP, Audio devices handling, > Start/stop RTP and encryption. I tried to find some documentation on openSBC > architecture but did not find much. Please give me some of your valuable > directions on how should i start with this. > > Regards, > > Manoj > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 4:59 PM > > > > |
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From: Manoj K. J. <ma...@as...> - 2008-08-12 14:36:07
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Thanks for Replying joegen. I am on it already. What about ... 1 - Starting Audio devices? 2- Start/Stop RTP I also need to understand how encryption is implemented as i would need to encrypt RTP also using same functions right? If i get an overall picture it would be real easy for me to device some flowchart on paper and proceed with work. Regards, Manoj -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Tuesday, August 12, 2008 6:57 PM To: ma...@as...; ope...@li... Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack The first thing you need to do is to implement your codec as a subclass of OpalFramedTranscoder. You can check out how Speex is implemented (speexcodec.h, speexcodec.cxx) and base you custom codec from there. The next step is to call your codec registration macro in allcodecs.h. Manoj Kumar Joshi wrote: > Hello, > > I am looking forward to integrate GIPS media processing to opensipstack. > Initially i want to incorporate it only on its Softphone interface and build > ATLSip with it. > > I think i would need to make changes in SDP, Audio devices handling, > Start/stop RTP and encryption. I tried to find some documentation on openSBC > architecture but did not find much. Please give me some of your valuable > directions on how should i start with this. > > Regards, > > Manoj > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 4:59 PM > > > > -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 7:23 PM |
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From: <jo...@op...> - 2008-08-12 23:57:18
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Manoj Kumar Joshi wrote: > Thanks for Replying joegen. I am on it already. What about ... > 1 - Starting Audio devices? > 2- Start/Stop RTP > Both these two should be seamless after you have successfully registered your codec. Unless, what you want is to rewrite everything and start from scratch? > I also need to understand how encryption is implemented as i would need to > encrypt RTP also using same functions right? If i get an overall picture it > would be real easy for me to device some flowchart on paper and proceed with > work. > Encryption is implemented in rtp.cxx and Encryption.cxx RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() BOOL RTP_UDP::WriteData() BOOL RTP_UDP::WriteControl() This too should be seamless after you introduced your codec. > Regards, > > Manoj > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Tuesday, August 12, 2008 6:57 PM > To: ma...@as...; ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > The first thing you need to do is to implement your codec as a subclass > of OpalFramedTranscoder. You can check out how Speex is implemented > (speexcodec.h, speexcodec.cxx) and base you custom codec from there. > The next step is to call your codec registration macro in allcodecs.h. > > Manoj Kumar Joshi wrote: > >> Hello, >> >> I am looking forward to integrate GIPS media processing to opensipstack. >> Initially i want to incorporate it only on its Softphone interface and >> > build > >> ATLSip with it. >> >> I think i would need to make changes in SDP, Audio devices handling, >> Start/stop RTP and encryption. I tried to find some documentation on >> > openSBC > >> architecture but did not find much. Please give me some of your valuable >> directions on how should i start with this. >> >> Regards, >> >> Manoj >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> > challenge > >> Build the coolest Linux based applications with Moblin SDK & win great >> > prizes > >> Grand prize is a trip for two to an Open Source event anywhere in the >> > world > >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> No virus found in this incoming message. >> Checked by AVG - http://www.avg.com >> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >> > 4:59 PM > >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 4:59 PM > > > > |
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From: Manoj K. J. <ma...@as...> - 2008-08-13 07:51:19
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Actually in my opinion i will have to do more than codec inclusion. I am not sure if you have seen GIPS functions..they way they use is... (When Softphone is Registered) - We create an instance of GIPS using Initialize function. (When a new call is initiated i.e. INVITE is about to sent) - We create a new channel - We specify RTP listen port (Same is sent in SIP SDP) (When Session progress comes) - We start listening to RTP port. - We start playing incoming media. - We set the "Send IP" and "Send port" (That comes in session progress SDP) (When 200 OK comes) - We start sending RTP to Mediaproxy IP and port (Hangup) - We close all channels. I have included a PDF file with this email. In page 31 there is a table which explains above points more. In Page 29 there is provision to add Encryption scheme. As i understand i might need to touch more than codec part in opensipstack but you will know better than me. -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Wednesday, August 13, 2008 5:27 AM To: ma...@as...; ope...@li... Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack Manoj Kumar Joshi wrote: > Thanks for Replying joegen. I am on it already. What about ... > 1 - Starting Audio devices? > 2- Start/Stop RTP > Both these two should be seamless after you have successfully registered your codec. Unless, what you want is to rewrite everything and start from scratch? > I also need to understand how encryption is implemented as i would need to > encrypt RTP also using same functions right? If i get an overall picture it > would be real easy for me to device some flowchart on paper and proceed with > work. > Encryption is implemented in rtp.cxx and Encryption.cxx RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() BOOL RTP_UDP::WriteData() BOOL RTP_UDP::WriteControl() This too should be seamless after you introduced your codec. > Regards, > > Manoj > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Tuesday, August 12, 2008 6:57 PM > To: ma...@as...; ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > The first thing you need to do is to implement your codec as a subclass > of OpalFramedTranscoder. You can check out how Speex is implemented > (speexcodec.h, speexcodec.cxx) and base you custom codec from there. > The next step is to call your codec registration macro in allcodecs.h. > > Manoj Kumar Joshi wrote: > >> Hello, >> >> I am looking forward to integrate GIPS media processing to opensipstack. >> Initially i want to incorporate it only on its Softphone interface and >> > build > >> ATLSip with it. >> >> I think i would need to make changes in SDP, Audio devices handling, >> Start/stop RTP and encryption. I tried to find some documentation on >> > openSBC > >> architecture but did not find much. Please give me some of your valuable >> directions on how should i start with this. >> >> Regards, >> >> Manoj >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> > challenge > >> Build the coolest Linux based applications with Moblin SDK & win great >> > prizes > >> Grand prize is a trip for two to an Open Source event anywhere in the >> > world > >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> No virus found in this incoming message. >> Checked by AVG - http://www.avg.com >> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >> > 4:59 PM > >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 4:59 PM > > > > -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 7:23 PM |
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From: <jo...@op...> - 2008-08-13 08:07:21
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Manoj Kumar Joshi wrote: > Actually in my opinion i will have to do more than codec inclusion. I am not > sure if you have seen GIPS functions..they way they use is... > > (When Softphone is Registered) > - We create an instance of GIPS using Initialize function. > (When a new call is initiated i.e. INVITE is about to sent) > - We create a new channel > - We specify RTP listen port (Same is sent in SIP SDP) > (When Session progress comes) > - We start listening to RTP port. > - We start playing incoming media. > - We set the "Send IP" and "Send port" (That comes in session progress SDP) > (When 200 OK comes) > - We start sending RTP to Mediaproxy IP and port > (Hangup) > - We close all channels. > As far as i can tell this will be done for you by the OpenSIPStack subsystem. Just register your codec to it. I do not know anything about GIPs but if its the same as all other codecs which basically has the ability to expose, encode and decode methods, then all you need is plug it in. > I have included a PDF file with this email. In page 31 there is a table > which explains above points more. In Page 29 there is provision to add > Encryption scheme. > > As i understand i might need to touch more than codec part in opensipstack > but you will know better than me. > > > I was under the impression that you are referring to the XOR encryption of opensipstack. If its another proprietary encryption other than srtp, then you will have to give more info how it works and probably I could point you to the right place where to hack it in OpenSIPStack. > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 5:27 AM > To: ma...@as...; ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > Manoj Kumar Joshi wrote: > >> Thanks for Replying joegen. I am on it already. What about ... >> 1 - Starting Audio devices? >> 2- Start/Stop RTP >> >> > > Both these two should be seamless after you have successfully registered > your codec. Unless, what you want is to rewrite everything and start > from scratch? > >> I also need to understand how encryption is implemented as i would need to >> encrypt RTP also using same functions right? If i get an overall picture >> > it > >> would be real easy for me to device some flowchart on paper and proceed >> > with > >> work. >> >> > > Encryption is implemented in rtp.cxx and Encryption.cxx > > RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() > BOOL RTP_UDP::WriteData() > BOOL RTP_UDP::WriteControl() > > This too should be seamless after you introduced your codec. > >> Regards, >> >> Manoj >> >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Tuesday, August 12, 2008 6:57 PM >> To: ma...@as...; ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> The first thing you need to do is to implement your codec as a subclass >> of OpalFramedTranscoder. You can check out how Speex is implemented >> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >> The next step is to call your codec registration macro in allcodecs.h. >> >> Manoj Kumar Joshi wrote: >> >> >>> Hello, >>> >>> I am looking forward to integrate GIPS media processing to opensipstack. >>> Initially i want to incorporate it only on its Softphone interface and >>> >>> >> build >> >> >>> ATLSip with it. >>> >>> I think i would need to make changes in SDP, Audio devices handling, >>> Start/stop RTP and encryption. I tried to find some documentation on >>> >>> >> openSBC >> >> >>> architecture but did not find much. Please give me some of your valuable >>> directions on how should i start with this. >>> >>> Regards, >>> >>> Manoj >>> >>> >>> ------------------------------------------------------------------------- >>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>> >>> >> challenge >> >> >>> Build the coolest Linux based applications with Moblin SDK & win great >>> >>> >> prizes >> >> >>> Grand prize is a trip for two to an Open Source event anywhere in the >>> >>> >> world >> >> >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> No virus found in this incoming message. >>> Checked by AVG - http://www.avg.com >>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >>> >>> >> 4:59 PM >> >> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> > challenge > >> Build the coolest Linux based applications with Moblin SDK & win great >> > prizes > >> Grand prize is a trip for two to an Open Source event anywhere in the >> > world > >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> No virus found in this incoming message. >> Checked by AVG - http://www.avg.com >> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >> > 4:59 PM > >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > |
|
From: Manoj K. J. <ma...@as...> - 2008-08-13 08:42:36
|
Dear Joegen, GIPS is not only a set of few codec rather it has its own media processing system. The functions i specified in last email need to be called from various points of opensipstack. If you can direct me from where all media processing is done i will be able to understand better. In addition to that please also tell me following... >From what places Encryption related functions are being called (Like enable encrytion, then encrypting and decrpting all SIP messages) Also if i understand correctly threre is some key involved in order to encrypt RTP. If you can plaese give me complete flow it will be a great help. Best Regards, Manoj -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Wednesday, August 13, 2008 1:37 PM To: ma...@as... Cc: ope...@li... Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack Manoj Kumar Joshi wrote: > Actually in my opinion i will have to do more than codec inclusion. I am not > sure if you have seen GIPS functions..they way they use is... > > (When Softphone is Registered) > - We create an instance of GIPS using Initialize function. > (When a new call is initiated i.e. INVITE is about to sent) > - We create a new channel > - We specify RTP listen port (Same is sent in SIP SDP) > (When Session progress comes) > - We start listening to RTP port. > - We start playing incoming media. > - We set the "Send IP" and "Send port" (That comes in session progress SDP) > (When 200 OK comes) > - We start sending RTP to Mediaproxy IP and port > (Hangup) > - We close all channels. > As far as i can tell this will be done for you by the OpenSIPStack subsystem. Just register your codec to it. I do not know anything about GIPs but if its the same as all other codecs which basically has the ability to expose, encode and decode methods, then all you need is plug it in. > I have included a PDF file with this email. In page 31 there is a table > which explains above points more. In Page 29 there is provision to add > Encryption scheme. > > As i understand i might need to touch more than codec part in opensipstack > but you will know better than me. > > > I was under the impression that you are referring to the XOR encryption of opensipstack. If its another proprietary encryption other than srtp, then you will have to give more info how it works and probably I could point you to the right place where to hack it in OpenSIPStack. > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 5:27 AM > To: ma...@as...; ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > Manoj Kumar Joshi wrote: > >> Thanks for Replying joegen. I am on it already. What about ... >> 1 - Starting Audio devices? >> 2- Start/Stop RTP >> >> > > Both these two should be seamless after you have successfully registered > your codec. Unless, what you want is to rewrite everything and start > from scratch? > >> I also need to understand how encryption is implemented as i would need to >> encrypt RTP also using same functions right? If i get an overall picture >> > it > >> would be real easy for me to device some flowchart on paper and proceed >> > with > >> work. >> >> > > Encryption is implemented in rtp.cxx and Encryption.cxx > > RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() > BOOL RTP_UDP::WriteData() > BOOL RTP_UDP::WriteControl() > > This too should be seamless after you introduced your codec. > >> Regards, >> >> Manoj >> >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Tuesday, August 12, 2008 6:57 PM >> To: ma...@as...; ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> The first thing you need to do is to implement your codec as a subclass >> of OpalFramedTranscoder. You can check out how Speex is implemented >> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >> The next step is to call your codec registration macro in allcodecs.h. >> >> Manoj Kumar Joshi wrote: >> >> >>> Hello, >>> >>> I am looking forward to integrate GIPS media processing to opensipstack. >>> Initially i want to incorporate it only on its Softphone interface and >>> >>> >> build >> >> >>> ATLSip with it. >>> >>> I think i would need to make changes in SDP, Audio devices handling, >>> Start/stop RTP and encryption. I tried to find some documentation on >>> >>> >> openSBC >> >> >>> architecture but did not find much. Please give me some of your valuable >>> directions on how should i start with this. >>> >>> Regards, >>> >>> Manoj >>> >>> >>> ------------------------------------------------------------------------ - >>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>> >>> >> challenge >> >> >>> Build the coolest Linux based applications with Moblin SDK & win great >>> >>> >> prizes >> >> >>> Grand prize is a trip for two to an Open Source event anywhere in the >>> >>> >> world >> >> >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> No virus found in this incoming message. >>> Checked by AVG - http://www.avg.com >>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >>> >>> >> 4:59 PM >> >> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> > challenge > >> Build the coolest Linux based applications with Moblin SDK & win great >> > prizes > >> Grand prize is a trip for two to an Open Source event anywhere in the >> > world > >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> No virus found in this incoming message. >> Checked by AVG - http://www.avg.com >> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >> > 4:59 PM > >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 7:23 PM |
|
From: <jo...@op...> - 2008-08-13 08:56:24
|
GIPS has its own RTP Stack, audio input/output channels? Manoj Kumar Joshi wrote: > Dear Joegen, > > GIPS is not only a set of few codec rather it has its own media processing > system. The functions i specified in last email need to be called from > various points of opensipstack. If you can direct me from where all media > processing is done i will be able to understand better. > > In addition to that please also tell me following... > > >From what places Encryption related functions are being called (Like enable > encrytion, then encrypting and decrpting all SIP messages) Also if i > understand correctly threre is some key involved in order to encrypt RTP. If > you can plaese give me complete flow it will be a great help. > > Best Regards, > > Manoj > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 1:37 PM > To: ma...@as... > Cc: ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > Manoj Kumar Joshi wrote: > >> Actually in my opinion i will have to do more than codec inclusion. I am >> > not > >> sure if you have seen GIPS functions..they way they use is... >> >> (When Softphone is Registered) >> - We create an instance of GIPS using Initialize function. >> (When a new call is initiated i.e. INVITE is about to sent) >> - We create a new channel >> - We specify RTP listen port (Same is sent in SIP SDP) >> (When Session progress comes) >> - We start listening to RTP port. >> - We start playing incoming media. >> - We set the "Send IP" and "Send port" (That comes in session progress >> > SDP) > >> (When 200 OK comes) >> - We start sending RTP to Mediaproxy IP and port >> (Hangup) >> - We close all channels. >> >> > > > As far as i can tell this will be done for you by the OpenSIPStack > subsystem. Just register your codec to it. I do not know anything > about GIPs but if its the same as all other codecs which basically has > the ability to expose, encode and decode methods, then all you need is > plug it in. > > > > >> I have included a PDF file with this email. In page 31 there is a table >> which explains above points more. In Page 29 there is provision to add >> Encryption scheme. >> >> As i understand i might need to touch more than codec part in opensipstack >> but you will know better than me. >> >> >> >> > > > I was under the impression that you are referring to the XOR encryption > of opensipstack. If its another proprietary encryption other than srtp, > then you will have to give more info how it works and probably I could > point you to the right place where to hack it in OpenSIPStack. > > > >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Wednesday, August 13, 2008 5:27 AM >> To: ma...@as...; ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> Manoj Kumar Joshi wrote: >> >> >>> Thanks for Replying joegen. I am on it already. What about ... >>> 1 - Starting Audio devices? >>> 2- Start/Stop RTP >>> >>> >>> >> Both these two should be seamless after you have successfully registered >> your codec. Unless, what you want is to rewrite everything and start >> from scratch? >> >> >>> I also need to understand how encryption is implemented as i would need >>> > to > >>> encrypt RTP also using same functions right? If i get an overall picture >>> >>> >> it >> >> >>> would be real easy for me to device some flowchart on paper and proceed >>> >>> >> with >> >> >>> work. >>> >>> >>> >> Encryption is implemented in rtp.cxx and Encryption.cxx >> >> RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() >> BOOL RTP_UDP::WriteData() >> BOOL RTP_UDP::WriteControl() >> >> This too should be seamless after you introduced your codec. >> >> >>> Regards, >>> >>> Manoj >>> >>> -----Original Message----- >>> From: jo...@op... [mailto:joe...@gm...] >>> Sent: Tuesday, August 12, 2008 6:57 PM >>> To: ma...@as...; ope...@li... >>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>> >>> >>> The first thing you need to do is to implement your codec as a subclass >>> of OpalFramedTranscoder. You can check out how Speex is implemented >>> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >>> The next step is to call your codec registration macro in allcodecs.h. >>> >>> Manoj Kumar Joshi wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I am looking forward to integrate GIPS media processing to opensipstack. >>>> Initially i want to incorporate it only on its Softphone interface and >>>> >>>> >>>> >>> build >>> >>> >>> >>>> ATLSip with it. >>>> >>>> I think i would need to make changes in SDP, Audio devices handling, >>>> Start/stop RTP and encryption. I tried to find some documentation on >>>> >>>> >>>> >>> openSBC >>> >>> >>> >>>> architecture but did not find much. Please give me some of your valuable >>>> directions on how should i start with this. >>>> >>>> Regards, >>>> >>>> Manoj >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> > - > >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>> >>>> >>>> >>> challenge >>> >>> >>> >>>> Build the coolest Linux based applications with Moblin SDK & win great >>>> >>>> >>>> >>> prizes >>> >>> >>> >>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>> >>>> >>>> >>> world >>> >>> >>> >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> No virus found in this incoming message. >>>> Checked by AVG - http://www.avg.com >>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>> > 8/11/2008 > >>>> >>> 4:59 PM >>> >>> >>> >>>> >>> -- >>> Internal Virus Database is out-of-date. >>> Checked by AVG. >>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >>> 7:23 PM >>> >>> >>> >>> ------------------------------------------------------------------------- >>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>> >>> >> challenge >> >> >>> Build the coolest Linux based applications with Moblin SDK & win great >>> >>> >> prizes >> >> >>> Grand prize is a trip for two to an Open Source event anywhere in the >>> >>> >> world >> >> >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> No virus found in this incoming message. >>> Checked by AVG - http://www.avg.com >>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >>> >>> >> 4:59 PM >> >> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > > |
|
From: Manoj K. J. <ma...@as...> - 2008-08-13 09:01:47
|
Yes -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Wednesday, August 13, 2008 2:26 PM To: ma...@as... Cc: ope...@li... Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack GIPS has its own RTP Stack, audio input/output channels? Manoj Kumar Joshi wrote: > Dear Joegen, > > GIPS is not only a set of few codec rather it has its own media processing > system. The functions i specified in last email need to be called from > various points of opensipstack. If you can direct me from where all media > processing is done i will be able to understand better. > > In addition to that please also tell me following... > > >From what places Encryption related functions are being called (Like enable > encrytion, then encrypting and decrpting all SIP messages) Also if i > understand correctly threre is some key involved in order to encrypt RTP. If > you can plaese give me complete flow it will be a great help. > > Best Regards, > > Manoj > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 1:37 PM > To: ma...@as... > Cc: ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > Manoj Kumar Joshi wrote: > >> Actually in my opinion i will have to do more than codec inclusion. I am >> > not > >> sure if you have seen GIPS functions..they way they use is... >> >> (When Softphone is Registered) >> - We create an instance of GIPS using Initialize function. >> (When a new call is initiated i.e. INVITE is about to sent) >> - We create a new channel >> - We specify RTP listen port (Same is sent in SIP SDP) >> (When Session progress comes) >> - We start listening to RTP port. >> - We start playing incoming media. >> - We set the "Send IP" and "Send port" (That comes in session progress >> > SDP) > >> (When 200 OK comes) >> - We start sending RTP to Mediaproxy IP and port >> (Hangup) >> - We close all channels. >> >> > > > As far as i can tell this will be done for you by the OpenSIPStack > subsystem. Just register your codec to it. I do not know anything > about GIPs but if its the same as all other codecs which basically has > the ability to expose, encode and decode methods, then all you need is > plug it in. > > > > >> I have included a PDF file with this email. In page 31 there is a table >> which explains above points more. In Page 29 there is provision to add >> Encryption scheme. >> >> As i understand i might need to touch more than codec part in opensipstack >> but you will know better than me. >> >> >> >> > > > I was under the impression that you are referring to the XOR encryption > of opensipstack. If its another proprietary encryption other than srtp, > then you will have to give more info how it works and probably I could > point you to the right place where to hack it in OpenSIPStack. > > > >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Wednesday, August 13, 2008 5:27 AM >> To: ma...@as...; ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> Manoj Kumar Joshi wrote: >> >> >>> Thanks for Replying joegen. I am on it already. What about ... >>> 1 - Starting Audio devices? >>> 2- Start/Stop RTP >>> >>> >>> >> Both these two should be seamless after you have successfully registered >> your codec. Unless, what you want is to rewrite everything and start >> from scratch? >> >> >>> I also need to understand how encryption is implemented as i would need >>> > to > >>> encrypt RTP also using same functions right? If i get an overall picture >>> >>> >> it >> >> >>> would be real easy for me to device some flowchart on paper and proceed >>> >>> >> with >> >> >>> work. >>> >>> >>> >> Encryption is implemented in rtp.cxx and Encryption.cxx >> >> RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() >> BOOL RTP_UDP::WriteData() >> BOOL RTP_UDP::WriteControl() >> >> This too should be seamless after you introduced your codec. >> >> >>> Regards, >>> >>> Manoj >>> >>> -----Original Message----- >>> From: jo...@op... [mailto:joe...@gm...] >>> Sent: Tuesday, August 12, 2008 6:57 PM >>> To: ma...@as...; ope...@li... >>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>> >>> >>> The first thing you need to do is to implement your codec as a subclass >>> of OpalFramedTranscoder. You can check out how Speex is implemented >>> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >>> The next step is to call your codec registration macro in allcodecs.h. >>> >>> Manoj Kumar Joshi wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I am looking forward to integrate GIPS media processing to opensipstack. >>>> Initially i want to incorporate it only on its Softphone interface and >>>> >>>> >>>> >>> build >>> >>> >>> >>>> ATLSip with it. >>>> >>>> I think i would need to make changes in SDP, Audio devices handling, >>>> Start/stop RTP and encryption. I tried to find some documentation on >>>> >>>> >>>> >>> openSBC >>> >>> >>> >>>> architecture but did not find much. Please give me some of your valuable >>>> directions on how should i start with this. >>>> >>>> Regards, >>>> >>>> Manoj >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>> >>>> >>>> >>> challenge >>> >>> >>> >>>> Build the coolest Linux based applications with Moblin SDK & win great >>>> >>>> >>>> >>> prizes >>> >>> >>> >>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>> >>>> >>>> >>> world >>> >>> >>> >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> No virus found in this incoming message. >>>> Checked by AVG - http://www.avg.com >>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>> > 8/11/2008 > >>>> >>> 4:59 PM >>> >>> >>> >>>> >>> -- >>> Internal Virus Database is out-of-date. >>> Checked by AVG. >>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >>> 7:23 PM >>> >>> >>> >>> ------------------------------------------------------------------------ - >>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>> >>> >> challenge >> >> >>> Build the coolest Linux based applications with Moblin SDK & win great >>> >>> >> prizes >> >> >>> Grand prize is a trip for two to an Open Source event anywhere in the >>> >>> >> world >> >> >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> No virus found in this incoming message. >>> Checked by AVG - http://www.avg.com >>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 8/11/2008 >>> >>> >> 4:59 PM >> >> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > > -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 7:23 PM |
|
From: <jo...@op...> - 2008-08-13 10:11:42
|
If GIPs has its own RTP and media input/output channels, would GIPS be willing to share it with other codecs as well if SDP negotiation ended up with a none-GIPS codec? If not, then you wouldn't be able to re-use the SoftPhone classes and instead implement your own subclass of CallSession and CallSessionManager. If GIPS RTP and Audio Channels are exposed to other codecs, then it would be doable by merely replacing RTP_UDPSession with the GIPS counterpart and a GIPS wrapper to PSoundChannel. Manoj Kumar Joshi wrote: > Yes > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 2:26 PM > To: ma...@as... > Cc: ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > GIPS has its own RTP Stack, audio input/output channels? > > Manoj Kumar Joshi wrote: > >> Dear Joegen, >> >> GIPS is not only a set of few codec rather it has its own media processing >> system. The functions i specified in last email need to be called from >> various points of opensipstack. If you can direct me from where all media >> processing is done i will be able to understand better. >> >> In addition to that please also tell me following... >> >> >From what places Encryption related functions are being called (Like >> > enable > >> encrytion, then encrypting and decrpting all SIP messages) Also if i >> understand correctly threre is some key involved in order to encrypt RTP. >> > If > >> you can plaese give me complete flow it will be a great help. >> >> Best Regards, >> >> Manoj >> >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Wednesday, August 13, 2008 1:37 PM >> To: ma...@as... >> Cc: ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> Manoj Kumar Joshi wrote: >> >> >>> Actually in my opinion i will have to do more than codec inclusion. I am >>> >>> >> not >> >> >>> sure if you have seen GIPS functions..they way they use is... >>> >>> (When Softphone is Registered) >>> - We create an instance of GIPS using Initialize function. >>> (When a new call is initiated i.e. INVITE is about to sent) >>> - We create a new channel >>> - We specify RTP listen port (Same is sent in SIP SDP) >>> (When Session progress comes) >>> - We start listening to RTP port. >>> - We start playing incoming media. >>> - We set the "Send IP" and "Send port" (That comes in session progress >>> >>> >> SDP) >> >> >>> (When 200 OK comes) >>> - We start sending RTP to Mediaproxy IP and port >>> (Hangup) >>> - We close all channels. >>> >>> >>> >> As far as i can tell this will be done for you by the OpenSIPStack >> subsystem. Just register your codec to it. I do not know anything >> about GIPs but if its the same as all other codecs which basically has >> the ability to expose, encode and decode methods, then all you need is >> plug it in. >> >> >> >> >> >>> I have included a PDF file with this email. In page 31 there is a table >>> which explains above points more. In Page 29 there is provision to add >>> Encryption scheme. >>> >>> As i understand i might need to touch more than codec part in >>> > opensipstack > >>> but you will know better than me. >>> >>> >>> >>> >>> >> I was under the impression that you are referring to the XOR encryption >> of opensipstack. If its another proprietary encryption other than srtp, >> then you will have to give more info how it works and probably I could >> point you to the right place where to hack it in OpenSIPStack. >> >> >> >> >>> -----Original Message----- >>> From: jo...@op... [mailto:joe...@gm...] >>> Sent: Wednesday, August 13, 2008 5:27 AM >>> To: ma...@as...; ope...@li... >>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>> >>> >>> Manoj Kumar Joshi wrote: >>> >>> >>> >>>> Thanks for Replying joegen. I am on it already. What about ... >>>> 1 - Starting Audio devices? >>>> 2- Start/Stop RTP >>>> >>>> >>>> >>>> >>> Both these two should be seamless after you have successfully registered >>> your codec. Unless, what you want is to rewrite everything and start >>> from scratch? >>> >>> >>> >>>> I also need to understand how encryption is implemented as i would need >>>> >>>> >> to >> >> >>>> encrypt RTP also using same functions right? If i get an overall picture >>>> >>>> >>>> >>> it >>> >>> >>> >>>> would be real easy for me to device some flowchart on paper and proceed >>>> >>>> >>>> >>> with >>> >>> >>> >>>> work. >>>> >>>> >>>> >>>> >>> Encryption is implemented in rtp.cxx and Encryption.cxx >>> >>> RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() >>> BOOL RTP_UDP::WriteData() >>> BOOL RTP_UDP::WriteControl() >>> >>> This too should be seamless after you introduced your codec. >>> >>> >>> >>>> Regards, >>>> >>>> Manoj >>>> >>>> -----Original Message----- >>>> From: jo...@op... [mailto:joe...@gm...] >>>> Sent: Tuesday, August 12, 2008 6:57 PM >>>> To: ma...@as...; ope...@li... >>>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>>> >>>> >>>> The first thing you need to do is to implement your codec as a subclass >>>> of OpalFramedTranscoder. You can check out how Speex is implemented >>>> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >>>> The next step is to call your codec registration macro in allcodecs.h. >>>> >>>> Manoj Kumar Joshi wrote: >>>> >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I am looking forward to integrate GIPS media processing to >>>>> > opensipstack. > >>>>> Initially i want to incorporate it only on its Softphone interface and >>>>> >>>>> >>>>> >>>>> >>>> build >>>> >>>> >>>> >>>> >>>>> ATLSip with it. >>>>> >>>>> I think i would need to make changes in SDP, Audio devices handling, >>>>> Start/stop RTP and encryption. I tried to find some documentation on >>>>> >>>>> >>>>> >>>>> >>>> openSBC >>>> >>>> >>>> >>>> >>>>> architecture but did not find much. Please give me some of your >>>>> > valuable > >>>>> directions on how should i start with this. >>>>> >>>>> Regards, >>>>> >>>>> Manoj >>>>> >>>>> >>>>> ----------------------------------------------------------------------- >>>>> > - > >> - >> >> >>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>> >>>>> >>>>> >>>>> >>>> challenge >>>> >>>> >>>> >>>> >>>>> Build the coolest Linux based applications with Moblin SDK & win great >>>>> >>>>> >>>>> >>>>> >>>> prizes >>>> >>>> >>>> >>>> >>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>> >>>>> >>>>> >>>>> >>>> world >>>> >>>> >>>> >>>> >>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>>> No virus found in this incoming message. >>>>> Checked by AVG - http://www.avg.com >>>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>>> >>>>> >> 8/11/2008 >> >> >>>> 4:59 PM >>>> >>>> >>>> >>>> >>>> -- >>>> Internal Virus Database is out-of-date. >>>> Checked by AVG. >>>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: >>>> > 8/4/2008 > >>>> 7:23 PM >>>> >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> > - > >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>> >>>> >>>> >>> challenge >>> >>> >>> >>>> Build the coolest Linux based applications with Moblin SDK & win great >>>> >>>> >>>> >>> prizes >>> >>> >>> >>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>> >>>> >>>> >>> world >>> >>> >>> >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> No virus found in this incoming message. >>>> Checked by AVG - http://www.avg.com >>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>> > 8/11/2008 > >>>> >>> 4:59 PM >>> >>> >>> >>>> >>> -- >>> Internal Virus Database is out-of-date. >>> Checked by AVG. >>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >>> 7:23 PM >>> >>> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1608 - Release Date: 8/12/2008 4:59 PM > > > > |
|
From: Manoj K. J. <ma...@as...> - 2008-08-13 10:22:04
|
I guess there is one way to use non-gips codecs also. I will do some more research and answer you. Please also answer following . >From what places Encryption related functions are being called (Like enable encrytion, then encrypting and decrpting all SIP messages) Also if i understand correctly threre is some key involved in order to encrypt RTP. Best Regards, Manoj -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Wednesday, August 13, 2008 3:42 PM To: ma...@as...; ope...@li... Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack If GIPs has its own RTP and media input/output channels, would GIPS be willing to share it with other codecs as well if SDP negotiation ended up with a none-GIPS codec? If not, then you wouldn't be able to re-use the SoftPhone classes and instead implement your own subclass of CallSession and CallSessionManager. If GIPS RTP and Audio Channels are exposed to other codecs, then it would be doable by merely replacing RTP_UDPSession with the GIPS counterpart and a GIPS wrapper to PSoundChannel. Manoj Kumar Joshi wrote: > Yes > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 2:26 PM > To: ma...@as... > Cc: ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > GIPS has its own RTP Stack, audio input/output channels? > > Manoj Kumar Joshi wrote: > >> Dear Joegen, >> >> GIPS is not only a set of few codec rather it has its own media processing >> system. The functions i specified in last email need to be called from >> various points of opensipstack. If you can direct me from where all media >> processing is done i will be able to understand better. >> >> In addition to that please also tell me following... >> >> >From what places Encryption related functions are being called (Like >> > enable > >> encrytion, then encrypting and decrpting all SIP messages) Also if i >> understand correctly threre is some key involved in order to encrypt RTP. >> > If > >> you can plaese give me complete flow it will be a great help. >> >> Best Regards, >> >> Manoj >> >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Wednesday, August 13, 2008 1:37 PM >> To: ma...@as... >> Cc: ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> Manoj Kumar Joshi wrote: >> >> >>> Actually in my opinion i will have to do more than codec inclusion. I am >>> >>> >> not >> >> >>> sure if you have seen GIPS functions..they way they use is... >>> >>> (When Softphone is Registered) >>> - We create an instance of GIPS using Initialize function. >>> (When a new call is initiated i.e. INVITE is about to sent) >>> - We create a new channel >>> - We specify RTP listen port (Same is sent in SIP SDP) >>> (When Session progress comes) >>> - We start listening to RTP port. >>> - We start playing incoming media. >>> - We set the "Send IP" and "Send port" (That comes in session progress >>> >>> >> SDP) >> >> >>> (When 200 OK comes) >>> - We start sending RTP to Mediaproxy IP and port >>> (Hangup) >>> - We close all channels. >>> >>> >>> >> As far as i can tell this will be done for you by the OpenSIPStack >> subsystem. Just register your codec to it. I do not know anything >> about GIPs but if its the same as all other codecs which basically has >> the ability to expose, encode and decode methods, then all you need is >> plug it in. >> >> >> >> >> >>> I have included a PDF file with this email. In page 31 there is a table >>> which explains above points more. In Page 29 there is provision to add >>> Encryption scheme. >>> >>> As i understand i might need to touch more than codec part in >>> > opensipstack > >>> but you will know better than me. >>> >>> >>> >>> >>> >> I was under the impression that you are referring to the XOR encryption >> of opensipstack. If its another proprietary encryption other than srtp, >> then you will have to give more info how it works and probably I could >> point you to the right place where to hack it in OpenSIPStack. >> >> >> >> >>> -----Original Message----- >>> From: jo...@op... [mailto:joe...@gm...] >>> Sent: Wednesday, August 13, 2008 5:27 AM >>> To: ma...@as...; ope...@li... >>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>> >>> >>> Manoj Kumar Joshi wrote: >>> >>> >>> >>>> Thanks for Replying joegen. I am on it already. What about ... >>>> 1 - Starting Audio devices? >>>> 2- Start/Stop RTP >>>> >>>> >>>> >>>> >>> Both these two should be seamless after you have successfully registered >>> your codec. Unless, what you want is to rewrite everything and start >>> from scratch? >>> >>> >>> >>>> I also need to understand how encryption is implemented as i would need >>>> >>>> >> to >> >> >>>> encrypt RTP also using same functions right? If i get an overall picture >>>> >>>> >>>> >>> it >>> >>> >>> >>>> would be real easy for me to device some flowchart on paper and proceed >>>> >>>> >>>> >>> with >>> >>> >>> >>>> work. >>>> >>>> >>>> >>>> >>> Encryption is implemented in rtp.cxx and Encryption.cxx >>> >>> RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() >>> BOOL RTP_UDP::WriteData() >>> BOOL RTP_UDP::WriteControl() >>> >>> This too should be seamless after you introduced your codec. >>> >>> >>> >>>> Regards, >>>> >>>> Manoj >>>> >>>> -----Original Message----- >>>> From: jo...@op... [mailto:joe...@gm...] >>>> Sent: Tuesday, August 12, 2008 6:57 PM >>>> To: ma...@as...; ope...@li... >>>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>>> >>>> >>>> The first thing you need to do is to implement your codec as a subclass >>>> of OpalFramedTranscoder. You can check out how Speex is implemented >>>> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >>>> The next step is to call your codec registration macro in allcodecs.h. >>>> >>>> Manoj Kumar Joshi wrote: >>>> >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I am looking forward to integrate GIPS media processing to >>>>> > opensipstack. > >>>>> Initially i want to incorporate it only on its Softphone interface and >>>>> >>>>> >>>>> >>>>> >>>> build >>>> >>>> >>>> >>>> >>>>> ATLSip with it. >>>>> >>>>> I think i would need to make changes in SDP, Audio devices handling, >>>>> Start/stop RTP and encryption. I tried to find some documentation on >>>>> >>>>> >>>>> >>>>> >>>> openSBC >>>> >>>> >>>> >>>> >>>>> architecture but did not find much. Please give me some of your >>>>> > valuable > >>>>> directions on how should i start with this. >>>>> >>>>> Regards, >>>>> >>>>> Manoj >>>>> >>>>> >>>>> ---------------------------------------------------------------------- - >>>>> > - > >> - >> >> >>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>> >>>>> >>>>> >>>>> >>>> challenge >>>> >>>> >>>> >>>> >>>>> Build the coolest Linux based applications with Moblin SDK & win great >>>>> >>>>> >>>>> >>>>> >>>> prizes >>>> >>>> >>>> >>>> >>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>> >>>>> >>>>> >>>>> >>>> world >>>> >>>> >>>> >>>> >>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>>> No virus found in this incoming message. >>>>> Checked by AVG - http://www.avg.com >>>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>>> >>>>> >> 8/11/2008 >> >> >>>> 4:59 PM >>>> >>>> >>>> >>>> >>>> -- >>>> Internal Virus Database is out-of-date. >>>> Checked by AVG. >>>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: >>>> > 8/4/2008 > >>>> 7:23 PM >>>> >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>> >>>> >>>> >>> challenge >>> >>> >>> >>>> Build the coolest Linux based applications with Moblin SDK & win great >>>> >>>> >>>> >>> prizes >>> >>> >>> >>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>> >>>> >>>> >>> world >>> >>> >>> >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> No virus found in this incoming message. >>>> Checked by AVG - http://www.avg.com >>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>> > 8/11/2008 > >>>> >>> 4:59 PM >>> >>> >>> >>>> >>> -- >>> Internal Virus Database is out-of-date. >>> Checked by AVG. >>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >>> 7:23 PM >>> >>> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1608 - Release Date: 8/12/2008 4:59 PM > > > > -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 7:23 PM |
|
From: <jo...@op...> - 2008-08-14 01:35:45
|
Firing up file search in visual studio for "Engine::Encrypt" C:\var\home\oss\devel\opensipstack\src\SIPMessage.cxx(8963): return Encryption::Engine::Encrypt( in, out ); C:\var\home\oss\devel\opensipstack\src\SIPUDPTransport.cxx(346): Encryption::Engine::Encrypt( in, out ); C:\var\home\oss\devel\opensipstack\src\opal\src\rtp\rtp.cxx(2201): Encryption::Engine::Encrypt( frame, out ); C:\var\home\oss\devel\opensipstack\src\opal\src\rtp\rtp.cxx(2244): Encryption::Engine::Encrypt( frame, out ); By process of elimination, I would say SIPUDPTransport is the place you should be looking at for SIP Encryption. The key for encryption is a static member variable of Encryption::Engine::m_Key. Just set it to the value you like and off you go. Manoj Kumar Joshi wrote: > I guess there is one way to use non-gips codecs also. I will do some more > research and answer you. Please also answer following . > >From what places Encryption related functions are being called (Like enable > encrytion, then encrypting and decrpting all SIP messages) Also if i > understand correctly threre is some key involved in order to encrypt RTP. > > Best Regards, > > Manoj > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, August 13, 2008 3:42 PM > To: ma...@as...; ope...@li... > Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack > > > If GIPs has its own RTP and media input/output channels, would GIPS be > willing to share it with other codecs as well if SDP negotiation ended > up with a none-GIPS codec? If not, then you wouldn't be able to re-use > the SoftPhone classes and instead implement your own subclass of > CallSession and CallSessionManager. If GIPS RTP and Audio Channels are > exposed to other codecs, then it would be doable by merely replacing > RTP_UDPSession with the GIPS counterpart and a GIPS wrapper to > PSoundChannel. > > > Manoj Kumar Joshi wrote: > >> Yes >> >> -----Original Message----- >> From: jo...@op... [mailto:joe...@gm...] >> Sent: Wednesday, August 13, 2008 2:26 PM >> To: ma...@as... >> Cc: ope...@li... >> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >> >> >> GIPS has its own RTP Stack, audio input/output channels? >> >> Manoj Kumar Joshi wrote: >> >> >>> Dear Joegen, >>> >>> GIPS is not only a set of few codec rather it has its own media >>> > processing > >>> system. The functions i specified in last email need to be called from >>> various points of opensipstack. If you can direct me from where all media >>> processing is done i will be able to understand better. >>> >>> In addition to that please also tell me following... >>> >>> >From what places Encryption related functions are being called (Like >>> >>> >> enable >> >> >>> encrytion, then encrypting and decrpting all SIP messages) Also if i >>> understand correctly threre is some key involved in order to encrypt RTP. >>> >>> >> If >> >> >>> you can plaese give me complete flow it will be a great help. >>> >>> Best Regards, >>> >>> Manoj >>> >>> -----Original Message----- >>> From: jo...@op... [mailto:joe...@gm...] >>> Sent: Wednesday, August 13, 2008 1:37 PM >>> To: ma...@as... >>> Cc: ope...@li... >>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>> >>> >>> Manoj Kumar Joshi wrote: >>> >>> >>> >>>> Actually in my opinion i will have to do more than codec inclusion. I am >>>> >>>> >>>> >>> not >>> >>> >>> >>>> sure if you have seen GIPS functions..they way they use is... >>>> >>>> (When Softphone is Registered) >>>> - We create an instance of GIPS using Initialize function. >>>> (When a new call is initiated i.e. INVITE is about to sent) >>>> - We create a new channel >>>> - We specify RTP listen port (Same is sent in SIP SDP) >>>> (When Session progress comes) >>>> - We start listening to RTP port. >>>> - We start playing incoming media. >>>> - We set the "Send IP" and "Send port" (That comes in session progress >>>> >>>> >>>> >>> SDP) >>> >>> >>> >>>> (When 200 OK comes) >>>> - We start sending RTP to Mediaproxy IP and port >>>> (Hangup) >>>> - We close all channels. >>>> >>>> >>>> >>>> >>> As far as i can tell this will be done for you by the OpenSIPStack >>> subsystem. Just register your codec to it. I do not know anything >>> about GIPs but if its the same as all other codecs which basically has >>> the ability to expose, encode and decode methods, then all you need is >>> plug it in. >>> >>> >>> >>> >>> >>> >>>> I have included a PDF file with this email. In page 31 there is a table >>>> which explains above points more. In Page 29 there is provision to add >>>> Encryption scheme. >>>> >>>> As i understand i might need to touch more than codec part in >>>> >>>> >> opensipstack >> >> >>>> but you will know better than me. >>>> >>>> >>>> >>>> >>>> >>>> >>> I was under the impression that you are referring to the XOR encryption >>> of opensipstack. If its another proprietary encryption other than srtp, >>> then you will have to give more info how it works and probably I could >>> point you to the right place where to hack it in OpenSIPStack. >>> >>> >>> >>> >>> >>>> -----Original Message----- >>>> From: jo...@op... [mailto:joe...@gm...] >>>> Sent: Wednesday, August 13, 2008 5:27 AM >>>> To: ma...@as...; ope...@li... >>>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>>> >>>> >>>> Manoj Kumar Joshi wrote: >>>> >>>> >>>> >>>> >>>>> Thanks for Replying joegen. I am on it already. What about ... >>>>> 1 - Starting Audio devices? >>>>> 2- Start/Stop RTP >>>>> >>>>> >>>>> >>>>> >>>>> >>>> Both these two should be seamless after you have successfully registered >>>> your codec. Unless, what you want is to rewrite everything and start >>>> from scratch? >>>> >>>> >>>> >>>> >>>>> I also need to understand how encryption is implemented as i would need >>>>> >>>>> >>>>> >>> to >>> >>> >>> >>>>> encrypt RTP also using same functions right? If i get an overall >>>>> > picture > >>>>> >>>>> >>>> it >>>> >>>> >>>> >>>> >>>>> would be real easy for me to device some flowchart on paper and proceed >>>>> >>>>> >>>>> >>>>> >>>> with >>>> >>>> >>>> >>>> >>>>> work. >>>>> >>>>> >>>>> >>>>> >>>>> >>>> Encryption is implemented in rtp.cxx and Encryption.cxx >>>> >>>> RTP_Session::SendReceiveStatus RTP_UDP::ReadDataOrControlPDU() >>>> BOOL RTP_UDP::WriteData() >>>> BOOL RTP_UDP::WriteControl() >>>> >>>> This too should be seamless after you introduced your codec. >>>> >>>> >>>> >>>> >>>>> Regards, >>>>> >>>>> Manoj >>>>> >>>>> -----Original Message----- >>>>> From: jo...@op... [mailto:joe...@gm...] >>>>> Sent: Tuesday, August 12, 2008 6:57 PM >>>>> To: ma...@as...; ope...@li... >>>>> Subject: Re: [OpenSIPStack] GIPs integration to OpenSIPStack >>>>> >>>>> >>>>> The first thing you need to do is to implement your codec as a subclass >>>>> of OpalFramedTranscoder. You can check out how Speex is implemented >>>>> (speexcodec.h, speexcodec.cxx) and base you custom codec from there. >>>>> The next step is to call your codec registration macro in allcodecs.h. >>>>> >>>>> Manoj Kumar Joshi wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Hello, >>>>>> >>>>>> I am looking forward to integrate GIPS media processing to >>>>>> >>>>>> >> opensipstack. >> >> >>>>>> Initially i want to incorporate it only on its Softphone interface and >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> build >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> ATLSip with it. >>>>>> >>>>>> I think i would need to make changes in SDP, Audio devices handling, >>>>>> Start/stop RTP and encryption. I tried to find some documentation on >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> openSBC >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> architecture but did not find much. Please give me some of your >>>>>> >>>>>> >> valuable >> >> >>>>>> directions on how should i start with this. >>>>>> >>>>>> Regards, >>>>>> >>>>>> Manoj >>>>>> >>>>>> >>>>>> ---------------------------------------------------------------------- >>>>>> > - > >> - >> >> >>> - >>> >>> >>> >>>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> challenge >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Build the coolest Linux based applications with Moblin SDK & win great >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> prizes >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> world >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> >>>>>> No virus found in this incoming message. >>>>>> Checked by AVG - http://www.avg.com >>>>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>>>> >>>>>> >>>>>> >>> 8/11/2008 >>> >>> >>> >>>>> 4:59 PM >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Internal Virus Database is out-of-date. >>>>> Checked by AVG. >>>>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: >>>>> >>>>> >> 8/4/2008 >> >> >>>>> 7:23 PM >>>>> >>>>> >>>>> >>>>> ----------------------------------------------------------------------- >>>>> > - > >> - >> >> >>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>> >>>>> >>>>> >>>>> >>>> challenge >>>> >>>> >>>> >>>> >>>>> Build the coolest Linux based applications with Moblin SDK & win great >>>>> >>>>> >>>>> >>>>> >>>> prizes >>>> >>>> >>>> >>>> >>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>> >>>>> >>>>> >>>>> >>>> world >>>> >>>> >>>> >>>> >>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>>> No virus found in this incoming message. >>>>> Checked by AVG - http://www.avg.com >>>>> Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: >>>>> >>>>> >> 8/11/2008 >> >> >>>> 4:59 PM >>>> >>>> >>>> >>>> >>>> -- >>>> Internal Virus Database is out-of-date. >>>> Checked by AVG. >>>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: >>>> > 8/4/2008 > >>>> 7:23 PM >>>> >>>> >>>> >>>> >>>> >>> -- >>> Internal Virus Database is out-of-date. >>> Checked by AVG. >>> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >>> 7:23 PM >>> >>> >>> >>> >>> >>> >>> >> >> -- >> Internal Virus Database is out-of-date. >> Checked by AVG. >> Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 >> 7:23 PM >> >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> > challenge > >> Build the coolest Linux based applications with Moblin SDK & win great >> > prizes > >> Grand prize is a trip for two to an Open Source event anywhere in the >> > world > >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> No virus found in this incoming message. >> Checked by AVG - http://www.avg.com >> Version: 8.0.138 / Virus Database: 270.6.1/1608 - Release Date: 8/12/2008 >> > 4:59 PM > >> >> >> > > > > -- > Internal Virus Database is out-of-date. > Checked by AVG. > Version: 7.5.524 / Virus Database: 270.5.12/1591 - Release Date: 8/4/2008 > 7:23 PM > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.1/1608 - Release Date: 8/12/2008 4:59 PM > > > > |
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From: Manoj K. J. <ma...@as...> - 2008-08-14 15:03:57
|
Hello Joegen, I had downloaded AtlSIP and opensipstack from http://www.opensipstack.org/downloads.html. AtlSIP and OpenSIPstack compiled successfully, but when i try to build OSSphone it gave me MSVCRT conflict. I managed to ignored these errors and built it but now it gives me run time error. Its quite possible i did some mistake. What is the directory order for these projects and what inputs Ossphone takes? Regards, Manoj |
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From: <jo...@op...> - 2008-08-15 00:22:19
|
You must give more info. Where is the compiler output? Manoj Kumar Joshi wrote: > Hello Joegen, > > I had downloaded AtlSIP and opensipstack from > http://www.opensipstack.org/downloads.html. AtlSIP and OpenSIPstack compiled > successfully, but when i try to build OSSphone it gave me MSVCRT conflict. I > managed to ignored these errors and built it but now it gives me run time > error. > > Its quite possible i did some mistake. What is the directory order for these > projects and what inputs Ossphone takes? > > Regards, > > Manoj > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.138 / Virus Database: 270.6.3/1610 - Release Date: 8/13/2008 4:14 PM > > > > |
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