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From: Joegen E. B. <joe...@gm...> - 2006-11-23 00:15:40
|
Hi, I tried compiling opensipstack from the download site and indeed ThreadPool.cxx is not inlcuded in the Visual C++ 8.0 project. Just include opensipstack/src/ThreadPool.cxx under the "Tools/Source Files" folder of your project and opensipstack/inlcude/ThreadPool.h in "Tools/Header Files" folder. That should fix the linker error in opensbc. Thanks for reporting this bug. Joegen Lidija Busic (ZG/ETK) wrote: > > > > Hello, > > > > I am using Visual C++ 8.0 compiler. > > ThreadPool.cxx is not compiled when compiling opensipstack library. > > > > What should I do to compile opensipstack properly? I have followed > instructions from the documentation web page. > > > > Thanks, Lidija > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Lidija B. \(ZG/ETK\) <lid...@er...> - 2006-11-22 09:36:58
|
=20 Hello, =20 I am using Visual C++ 8.0 compiler.=20 ThreadPool.cxx is not compiled when compiling opensipstack library. =20 What should I do to compile opensipstack properly? I have followed instructions from the documentation web page. =20 Thanks, Lidija |
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From: Joegen E. B. <joe...@gm...> - 2006-11-21 23:38:28
|
Akin, You may download the ATLSIP code in CVS. It goes with the full source code of OSSPhone. Joegen Akin Ocal wrote: > Can you send me any registration code sample , i want just to register > to my sip account then call a uri in the same network , but there is > only "call" sample.... > > Thank you very much... > > _________________________________________________________________ > Spam filtresi ile virüslere karsi en güvenilir koruma, MSN PC > Koruma'dan geçer. http://www.msn.com.tr/security/ > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Akin O. <aki...@ho...> - 2006-11-21 21:53:08
|
Can you send me any registration code sample , i want just to register to my sip account then call a uri in the same network , but there is only "call" sample.... Thank you very much... _________________________________________________________________ Spam filtresi ile virüslere karsi en güvenilir koruma, MSN PC Koruma'dan geçer. http://www.msn.com.tr/security/ |
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From: Joegen E. B. <joe...@gm...> - 2006-11-21 00:33:27
|
Akin Ocal wrote: > I want to register to my voipbuster account , its uri is just > ak...@si... > > How should i use OSSPhone -> > > what should i use for user id ? your user Id should be akhin > what should i use for account id ? Your account ID should be your Authentication account... this may also be "akhin" depending on authentication configuration of your registrar > what should i use for registrar ? This should be sip1.voipbuster.com. This may also be your proxy. > > Thank you very much > > _________________________________________________________________ > Spam filtresi ile virüslere karsi en güvenilir koruma, MSN PC > Koruma'dan geçer. http://www.msn.com.tr/security/ > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Akin O. <aki...@ho...> - 2006-11-21 00:17:23
|
I want to register to my voipbuster account , its uri is just ak...@si... How should i use OSSPhone -> what should i use for user id ? what should i use for account id ? what should i use for registrar ? Thank you very much _________________________________________________________________ Spam filtresi ile virüslere karsi en güvenilir koruma, MSN PC Koruma'dan geçer. http://www.msn.com.tr/security/ |
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From: Joegen E. B. <joe...@gm...> - 2006-11-17 14:59:35
|
Hi, What compiler version are you using? Can you confirm if ThreadPool.cxx is getting compiled when compiling the opensipstack library? Lidija Busic (ZG/ETK) wrote: > > Hello, > > I am trying to build OpenSBC, but keep getting the following: > > ------ Build started: Project: OpenSBC, Configuration: Debug Win32 ------ > > Compiling... > > SBCRoutingHandler.cxx > > SBCCallHandler.cxx > > SBCAuthHandler.cxx > > RouteRecord.cxx > > Router.cxx > > OpenSBC.cxx > > Main.cxx > > Generating Code... > > Compiling manifest to resources... > > Linking... > > opensipstackd.lib(B2BMediaInterface.obj) : error LNK2019: unresolved > external symbol "public: __thiscall > Tools::ThreadPool::ThreadPool(int,int,enum PThread::Priority,class > PString const &)" > (??0ThreadPool@Tools@@QAE@HHW4Priority@PThread@@ABVPString@@@Z) > referenced in function "public: int __thiscall > B2BUA::B2BMediaInterface::CreateMediaAggregation(class RTP_UDP *,class > RTP_UDP *)" > (?CreateMediaAggregation@B2BMediaInterface@B2BUA@@QAEHPAVRTP_UDP@@0@Z) > > opensipstackd.lib(B2BMediaInterface.obj) : error LNK2019: unresolved > external symbol "public: int __thiscall > Tools::ThreadPool::DoWork(class PNotifier const &,int)" > (?DoWork@ThreadPool@Tools@@QAEHABVPNotifier@@H@Z) referenced in > function "public: int __thiscall > B2BUA::B2BMediaInterface::ProcessMediaAggregation(void)" > (?ProcessMediaAggregation@B2BMediaInterface@B2BUA@@QAEHXZ) > > Debug/OpenSBC.exe : fatal error LNK1120: 2 unresolved externals > > Build log was saved at "file://c:\OpenSIP\opensbc\Debug\BuildLog.htm" > > OpenSBC - 3 error(s), 0 warning(s) > > ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped > ========== > > I don’t know what I’m doing wrong. It seems that the problem is in > OpenSIPStack (I have version 1.1.3), but didn’t get any errors while > building it. > > I tried using the version of OpenSBC from the OpenSIPStack page and > the one from the CVS, but reported errors are the same. > > Thank you for your help. > > Lidija > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Lidija B. \(ZG/ETK\) <lid...@er...> - 2006-11-17 14:42:36
|
Hello, =20 I am trying to build OpenSBC, but keep getting the following: =20 ------ Build started: Project: OpenSBC, Configuration: Debug Win32 ------ Compiling... SBCRoutingHandler.cxx SBCCallHandler.cxx SBCAuthHandler.cxx RouteRecord.cxx Router.cxx OpenSBC.cxx Main.cxx Generating Code... Compiling manifest to resources... Linking... opensipstackd.lib(B2BMediaInterface.obj) : error LNK2019: unresolved external symbol "public: __thiscall Tools::ThreadPool::ThreadPool(int,int,enum PThread::Priority,class PString const &)" (??0ThreadPool@Tools@@QAE@HHW4Priority@PThread@@ABVPString@@@Z) referenced in function "public: int __thiscall B2BUA::B2BMediaInterface::CreateMediaAggregation(class RTP_UDP *,class RTP_UDP *)" (?CreateMediaAggregation@B2BMediaInterface@B2BUA@@QAEHPAVRTP_UDP@@0@Z) opensipstackd.lib(B2BMediaInterface.obj) : error LNK2019: unresolved external symbol "public: int __thiscall Tools::ThreadPool::DoWork(class PNotifier const &,int)" (?DoWork@ThreadPool@Tools@@QAEHABVPNotifier@@H@Z) referenced in function "public: int __thiscall B2BUA::B2BMediaInterface::ProcessMediaAggregation(void)" (?ProcessMediaAggregation@B2BMediaInterface@B2BUA@@QAEHXZ) Debug/OpenSBC.exe : fatal error LNK1120: 2 unresolved externals Build log was saved at "file://c:\OpenSIP\opensbc\Debug\BuildLog.htm" OpenSBC - 3 error(s), 0 warning(s) =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D Build: 0 succeeded, 1 failed, 0 = up-to-date, 0 skipped =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D =20 I don't know what I'm doing wrong. It seems that the problem is in OpenSIPStack (I have version 1.1.3), but didn't get any errors while building it. I tried using the version of OpenSBC from the OpenSIPStack page and the one from the CVS, but reported errors are the same. =20 Thank you for your help. Lidija =20 |
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From: Ji Z. <zha...@ho...> - 2006-11-17 13:56:26
|
Hi Joegen, Pretty clear - thanks very much for your help. Let me get more details about SIP and hopefully next time I'll come up with more intelligent questions to the list. ;-p Best regards, Ji > There are four types of transactions defined by RFC 3261. > > ICT - Invite Client Transaction > IST - Invite Server Transaction > NICT - None Invite Client Transaction > NIST - None Invite Server Transaction. > > > When an INVITE is sent by the UAC, the transaction created is an Invite > Client Transaction. This would be composed of the Initial INVITE request > that created the transaction, provisional responses like 100 Trying and > 180 Ringing, the final response 200 Ok and the ACK. To illustrate the > call flow... > > UAC - (ICT) UAS - (IST) > INVITE --------------> | > 100 Trying <-------- | > 180 Ringing <-------- | > 200 Ok <------------- | > ACK --------------> | > <---------CALL---------> > UAC - (NICT) UAS -(NIST) > BYE --------------> | > 200 Ok <------------- | > > The entire Invite message exchange up to the ACK compose a transaction. > None Invite transactions (like BYE) do not require to be ACKed. In the > example above the entire session is composed of two transactions. > > Ji Zhang wrote: >> Dear Joegen, >> >> This is an excellent explanation! Thank you so much, as well as for the >> mailing-list you recommended! Before I get more details from RFC 3261, >> could you help me to get clearer about the relationship of the following >> words you mentioned: socket, session and transaction? >> >> According to you comments, may I understand as follows: 1 socket may >> multiplex across n sessions; and in 1 session there may be n >> transactions? To cite the example you mentioned: >> >> ................................................. >>>> caller INVITEs someone >> << someone says 200 OK >>>> caller says ACK >> .................................................. >>>> someone or caller sends BYE >> <<someone or caller sends 200 OK >> .................................................. >> >> I suppose this belongs to 1 session. Is that right? And how many >> transactions are there? I know I can find the answer in RFC 3261, but if >> I could have the general understanding beforehand it would be superb. >> >> Thanks again, >> Ji >> >> >> >>> Hi Ji Zhang, >>> >>> inline >>> >>> Ji Zhang wrote: >>>> Dear all, >>>> >>>> I'm new to SIP and this list and I wish I could find some help from >>>> here. >>> Certainly this is not the wrong place to ask Q's about SIP. However >>> there are better places to ask Q's about SIP in general . I am >>> recommending sip...@cs.... opensipstack-devel is >>> not the wrong place either so most definitely you can ask SIP related >>> questions here. >>> >>>> Could anybody tell me how a SIP application distinguish its own >>>> session flow from others if there are more than one application >>>> sessions going on at the same time? >>> The best answer would of course be read RFC 3261. However here is a >>> brief answer. Each request that opens a transaction should have a >>> "Call-Id" header present in the request. This header should contain a >>> globally unique identifier for the session that the transaction belongs >>> to. Since a session maybe composed of several transactions, there are >>> several ways explained in RFC 3261 to distinguish between different >>> transactions occurring/belonging to a SIP session. A basic call setup >>> for example would be composed of an INVITE, 200 Ok response, ACK, BYE, >>> 200 Ok response. (each SIP message separated by a comma ) >>> >>> .................................................. >>> >> caller INVITEs someone >>> << someone says 200 OK >>> >> caller says ACK >>> ................................................... >>> >>someone or caller sends BYE >>> <<someone or caller sends 200 OK >>> ................................................... >>> >>> Since the INVITE and BYE would share a common Call-Id, you should check >>> other headers (enumerated below) to pair responses with the >>> corresponding requests. >>> >>> *Call-Id >>> 1. From tag >>> 2. Via Branch >>> 3. CSeq method. >>> 4. CSeq number. >>> 5. To Tag. >>> >>> >>>> Is it based on the socket level port number or the call-id in the SIP >>>> messages? >>> >>> Most definitely not. Call-Id as I have hinted above only identifies a >>> Call or a Session. A session may be composed of one or more >>> transactions. Request response pairing in SIP is not based on call-id >>> alone but a combination of the value of the headers I have enumerated >>> above. >>> >>> If you browse through the opensipstack code, you would see that >>> opensipstack uses a single socket (per network interface) to multiplex >>> across multiple sessions. Port is NEVER a basis for identifying >>> sessions in SIP even if you use connection oriented protocol like TCP as >>> the transport. >>>> >>>> I'm also trying to understand how SIP interact with socket. Is SIP >>>> working like FTP that using a well-known port number (say 5060) to >>>> establish a session and negotiate new port numbers to use for the >>>> following data transfer? >>> >>> If you are using TCP as the transport then you can have a listener and >>> "accept" incoming connections using a new TCP socket. However, this >>> approach is known to exhaust resources far too quickly. There is a >>> draft discussing how to "reuse" TCP connections across multiple sessions >>> (www.ietf.org/internet-drafts/*draft-ietf-sip-connect-reuse-07*.html). >>> >>>> I'd be very appreciated if someone could show me some light or point >>>> me to a good reference to help me understand. >>>> >>> >>> NO. SIP sessions are NEVER associated with the ports that received the >>> request that created the initial transaction for the session. >>> >>>> Many thanks, >>>> Ji >>>> ------------------------------------------------------------------------ >>>> >>>> >>>> ------------------------------------------------------------------------- >>>> >>>> Take Surveys. Earn Cash. Influence the Future of IT >>>> Join SourceForge.net's Techsay panel and you'll get the chance to share >>>> your >>>> opinions on IT & business topics through brief surveys - and earn cash >>>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >>>> ------------------------------------------------------------------------ >>>> >>>> >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>> >>> >>> ------------------------------------------------------------------------- >>> >>> Take Surveys. Earn Cash. Influence the Future of IT >>> Join SourceForge.net's Techsay panel and you'll get the chance to share >>> your >>> opinions on IT & business topics through brief surveys - and earn cash >>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >> >> > > |
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From: Joegen E. B. <joe...@gm...> - 2006-11-17 13:35:15
|
There are four types of transactions defined by RFC 3261. ICT - Invite Client Transaction IST - Invite Server Transaction NICT - None Invite Client Transaction NIST - None Invite Server Transaction. When an INVITE is sent by the UAC, the transaction created is an Invite Client Transaction. This would be composed of the Initial INVITE request that created the transaction, provisional responses like 100 Trying and 180 Ringing, the final response 200 Ok and the ACK. To illustrate the call flow... UAC - (ICT) UAS - (IST) INVITE --------------> | 100 Trying <-------- | 180 Ringing <-------- | 200 Ok <------------- | ACK --------------> | <---------CALL---------> UAC - (NICT) UAS -(NIST) BYE --------------> | 200 Ok <------------- | The entire Invite message exchange up to the ACK compose a transaction. None Invite transactions (like BYE) do not require to be ACKed. In the example above the entire session is composed of two transactions. Ji Zhang wrote: > Dear Joegen, > > This is an excellent explanation! Thank you so much, as well as for > the mailing-list you recommended! Before I get more details from RFC > 3261, could you help me to get clearer about the relationship of the > following words you mentioned: socket, session and transaction? > > According to you comments, may I understand as follows: 1 socket may > multiplex across n sessions; and in 1 session there may be n > transactions? To cite the example you mentioned: > > ................................................. >>> caller INVITEs someone > << someone says 200 OK >>> caller says ACK > .................................................. >>> someone or caller sends BYE > <<someone or caller sends 200 OK > .................................................. > > I suppose this belongs to 1 session. Is that right? And how many > transactions are there? I know I can find the answer in RFC 3261, but > if I could have the general understanding beforehand it would be superb. > > Thanks again, > Ji > > > >> Hi Ji Zhang, >> >> inline >> >> Ji Zhang wrote: >>> Dear all, >>> >>> I'm new to SIP and this list and I wish I could find some help from >>> here. >> Certainly this is not the wrong place to ask Q's about SIP. However >> there are better places to ask Q's about SIP in general . I am >> recommending sip...@cs.... opensipstack-devel is >> not the wrong place either so most definitely you can ask SIP related >> questions here. >> >>> Could anybody tell me how a SIP application distinguish its own >>> session flow from others if there are more than one application >>> sessions going on at the same time? >> The best answer would of course be read RFC 3261. However here is a >> brief answer. Each request that opens a transaction should have a >> "Call-Id" header present in the request. This header should contain a >> globally unique identifier for the session that the transaction belongs >> to. Since a session maybe composed of several transactions, there are >> several ways explained in RFC 3261 to distinguish between different >> transactions occurring/belonging to a SIP session. A basic call setup >> for example would be composed of an INVITE, 200 Ok response, ACK, BYE, >> 200 Ok response. (each SIP message separated by a comma ) >> >> .................................................. >> >> caller INVITEs someone >> << someone says 200 OK >> >> caller says ACK >> ................................................... >> >>someone or caller sends BYE >> <<someone or caller sends 200 OK >> ................................................... >> >> Since the INVITE and BYE would share a common Call-Id, you should check >> other headers (enumerated below) to pair responses with the >> corresponding requests. >> >> *Call-Id >> 1. From tag >> 2. Via Branch >> 3. CSeq method. >> 4. CSeq number. >> 5. To Tag. >> >> >>> Is it based on the socket level port number or the call-id in the SIP >>> messages? >> >> Most definitely not. Call-Id as I have hinted above only identifies a >> Call or a Session. A session may be composed of one or more >> transactions. Request response pairing in SIP is not based on call-id >> alone but a combination of the value of the headers I have enumerated >> above. >> >> If you browse through the opensipstack code, you would see that >> opensipstack uses a single socket (per network interface) to multiplex >> across multiple sessions. Port is NEVER a basis for identifying >> sessions in SIP even if you use connection oriented protocol like TCP as >> the transport. >>> >>> I'm also trying to understand how SIP interact with socket. Is SIP >>> working like FTP that using a well-known port number (say 5060) to >>> establish a session and negotiate new port numbers to use for the >>> following data transfer? >> >> If you are using TCP as the transport then you can have a listener and >> "accept" incoming connections using a new TCP socket. However, this >> approach is known to exhaust resources far too quickly. There is a >> draft discussing how to "reuse" TCP connections across multiple sessions >> (www.ietf.org/internet-drafts/*draft-ietf-sip-connect-reuse-07*.html). >> >>> I'd be very appreciated if someone could show me some light or point >>> me to a good reference to help me understand. >>> >> >> NO. SIP sessions are NEVER associated with the ports that received the >> request that created the initial transaction for the session. >> >>> Many thanks, >>> Ji >>> ------------------------------------------------------------------------ >>> >>> >>> ------------------------------------------------------------------------- >>> >>> Take Surveys. Earn Cash. Influence the Future of IT >>> Join SourceForge.net's Techsay panel and you'll get the chance to >>> share your >>> opinions on IT & business topics through brief surveys - and earn cash >>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >>> >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >> >> >> ------------------------------------------------------------------------- >> >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> share your >> opinions on IT & business topics through brief surveys - and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > |
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From: Ji Z. <zha...@ho...> - 2006-11-16 15:58:03
|
Dear Joegen, This is an excellent explanation! Thank you so much, as well as for the mailing-list you recommended! Before I get more details from RFC 3261, could you help me to get clearer about the relationship of the following words you mentioned: socket, session and transaction? According to you comments, may I understand as follows: 1 socket may multiplex across n sessions; and in 1 session there may be n transactions? To cite the example you mentioned: ................................................. >> caller INVITEs someone << someone says 200 OK >> caller says ACK .................................................. >>someone or caller sends BYE <<someone or caller sends 200 OK .................................................. I suppose this belongs to 1 session. Is that right? And how many transactions are there? I know I can find the answer in RFC 3261, but if I could have the general understanding beforehand it would be superb. Thanks again, Ji > Hi Ji Zhang, > > inline > > Ji Zhang wrote: >> Dear all, >> >> I'm new to SIP and this list and I wish I could find some help from here. > Certainly this is not the wrong place to ask Q's about SIP. However > there are better places to ask Q's about SIP in general . I am > recommending sip...@cs.... opensipstack-devel is > not the wrong place either so most definitely you can ask SIP related > questions here. > >> Could anybody tell me how a SIP application distinguish its own >> session flow from others if there are more than one application >> sessions going on at the same time? > The best answer would of course be read RFC 3261. However here is a > brief answer. Each request that opens a transaction should have a > "Call-Id" header present in the request. This header should contain a > globally unique identifier for the session that the transaction belongs > to. Since a session maybe composed of several transactions, there are > several ways explained in RFC 3261 to distinguish between different > transactions occurring/belonging to a SIP session. A basic call setup > for example would be composed of an INVITE, 200 Ok response, ACK, BYE, > 200 Ok response. (each SIP message separated by a comma ) > > .................................................. > >> caller INVITEs someone > << someone says 200 OK > >> caller says ACK > ................................................... > >>someone or caller sends BYE > <<someone or caller sends 200 OK > ................................................... > > Since the INVITE and BYE would share a common Call-Id, you should check > other headers (enumerated below) to pair responses with the > corresponding requests. > > *Call-Id > 1. From tag > 2. Via Branch > 3. CSeq method. > 4. CSeq number. > 5. To Tag. > > >> Is it based on the socket level port number or the call-id in the SIP >> messages? > > Most definitely not. Call-Id as I have hinted above only identifies a > Call or a Session. A session may be composed of one or more > transactions. Request response pairing in SIP is not based on call-id > alone but a combination of the value of the headers I have enumerated > above. > > If you browse through the opensipstack code, you would see that > opensipstack uses a single socket (per network interface) to multiplex > across multiple sessions. Port is NEVER a basis for identifying > sessions in SIP even if you use connection oriented protocol like TCP as > the transport. >> >> I'm also trying to understand how SIP interact with socket. Is SIP >> working like FTP that using a well-known port number (say 5060) to >> establish a session and negotiate new port numbers to use for the >> following data transfer? > > If you are using TCP as the transport then you can have a listener and > "accept" incoming connections using a new TCP socket. However, this > approach is known to exhaust resources far too quickly. There is a > draft discussing how to "reuse" TCP connections across multiple sessions > (www.ietf.org/internet-drafts/*draft-ietf-sip-connect-reuse-07*.html). > >> I'd be very appreciated if someone could show me some light or point >> me to a good reference to help me understand. >> > > NO. SIP sessions are NEVER associated with the ports that received the > request that created the initial transaction for the session. > >> Many thanks, >> Ji >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to share >> your >> opinions on IT & business topics through brief surveys - and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Joegen E. B. <joe...@gm...> - 2006-11-16 14:58:00
|
Hi Ji Zhang, inline Ji Zhang wrote: > Dear all, > > I'm new to SIP and this list and I wish I could find some help from here. Certainly this is not the wrong place to ask Q's about SIP. However there are better places to ask Q's about SIP in general . I am recommending sip...@cs.... opensipstack-devel is not the wrong place either so most definitely you can ask SIP related questions here. > Could anybody tell me how a SIP application distinguish its own > session flow from others if there are more than one application > sessions going on at the same time? The best answer would of course be read RFC 3261. However here is a brief answer. Each request that opens a transaction should have a "Call-Id" header present in the request. This header should contain a globally unique identifier for the session that the transaction belongs to. Since a session maybe composed of several transactions, there are several ways explained in RFC 3261 to distinguish between different transactions occurring/belonging to a SIP session. A basic call setup for example would be composed of an INVITE, 200 Ok response, ACK, BYE, 200 Ok response. (each SIP message separated by a comma ) .................................................. >> caller INVITEs someone << someone says 200 OK >> caller says ACK ................................................... >>someone or caller sends BYE <<someone or caller sends 200 OK ................................................... Since the INVITE and BYE would share a common Call-Id, you should check other headers (enumerated below) to pair responses with the corresponding requests. *Call-Id 1. From tag 2. Via Branch 3. CSeq method. 4. CSeq number. 5. To Tag. > Is it based on the socket level port number or the call-id in the SIP > messages? Most definitely not. Call-Id as I have hinted above only identifies a Call or a Session. A session may be composed of one or more transactions. Request response pairing in SIP is not based on call-id alone but a combination of the value of the headers I have enumerated above. If you browse through the opensipstack code, you would see that opensipstack uses a single socket (per network interface) to multiplex across multiple sessions. Port is NEVER a basis for identifying sessions in SIP even if you use connection oriented protocol like TCP as the transport. > > I'm also trying to understand how SIP interact with socket. Is SIP > working like FTP that using a well-known port number (say 5060) to > establish a session and negotiate new port numbers to use for the > following data transfer? If you are using TCP as the transport then you can have a listener and "accept" incoming connections using a new TCP socket. However, this approach is known to exhaust resources far too quickly. There is a draft discussing how to "reuse" TCP connections across multiple sessions (www.ietf.org/internet-drafts/*draft-ietf-sip-connect-reuse-07*.html). > I'd be very appreciated if someone could show me some light or point > me to a good reference to help me understand. > NO. SIP sessions are NEVER associated with the ports that received the request that created the initial transaction for the session. > Many thanks, > Ji > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Ji Z. <zha...@ho...> - 2006-11-16 12:08:36
|
Dear all, I'm new to SIP and this list and I wish I could find some help from = here. Could anybody tell me how a SIP application distinguish its own = session flow from others if there are more than one application sessions = going on at the same time? Is it based on the socket level port number = or the call-id in the SIP messages? I'm also trying to understand how SIP interact with socket. Is SIP = working like FTP that using a well-known port number (say 5060) to = establish a session and negotiate new port numbers to use for the = following data transfer? I'd be very appreciated if someone could show = me some light or point me to a good reference to help me understand. Many thanks, Ji |
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From: Joegen E. B. <joe...@gm...> - 2006-11-15 11:50:12
|
Hi Julio, I think I can spare some time preparing something in the documentation wiki for you. I'll let you know. Is there anything in particular you want to know about the roadmap and supported RFCs? Joegen Julio Cesar Esteves Cabezas wrote: > > Hi, > > > > Will be published some more detailed list of features of OSS, like > RFCs implemented, etc. ? > > > > Is there any foreseen roadmap for future developments of OSS, in > signalling (SIP), network and media resources (RTP, codecs, etc) ? > > > > Regards, > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Julio C. E. C. <jca...@in...> - 2006-11-14 20:42:04
|
Hi,
=20
Will be published some more detailed list of features of OSS, like RFCs
implemented, etc. ?
=20
Is there any foreseen roadmap for future developments of OSS, in
signalling (SIP), network and media resources (RTP, codecs, etc) ?
=20
Regards,
=20
|
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From: Joegen E. B. <jo...@pl...> - 2006-11-08 13:58:35
|
Leandro Caetano Gon=E7alves Lustosa wrote: > > Do we have access to the source? > > =20 You bet. It goes with the ATLSIP source tree. Currently only=20 available via CVS. |
|
From:
<lca...@gm...> - 2006-11-08 13:55:02
|
Hi Joegen, Pretty cool! I'm using OSSPhone with OpenSER and Asterisk successfully! I could noticed Digest autentication and SRV DNS queries work fine as well. Do we have access to the source? Leandro On 11/8/06, Joegen E. Baclor <joe...@gm...> wrote: > Hi All, > > I have uploaded a windows installer for OSSPhone for those who want to > try it out ( > http://www.opensipstack.org/public/downloads/OSSPhoneSetup.msi ). > Nothing fancy yet, but it will soon get a new face lift so stay tuned. > SIMPLE presence support is soon to be released some time in December! > > Joegen > > ------------------------------------------------------------------------- > Using Tomcat but need to do more? Need to support web services, security? > Get stuff done quickly with pre-integrated technology to make your job easier > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > -- Leandro |
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From: Joegen E. B. <joe...@gm...> - 2006-11-08 12:58:03
|
Hi All, I have uploaded a windows installer for OSSPhone for those who want to try it out ( http://www.opensipstack.org/public/downloads/OSSPhoneSetup.msi ). Nothing fancy yet, but it will soon get a new face lift so stay tuned. SIMPLE presence support is soon to be released some time in December! Joegen |
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From: Joegen E. B. <joe...@gm...> - 2006-11-07 23:30:08
|
Leandro,
No need to panic. This isn't a real bug in a sense. Visual C++ 2005
compiler has implemented strict variable scoping for "for loops".
Variables declared within the for loop argument list like "i" in this
instance is only visible within the for loop code block. Your fix
should be fine. This has already been corrected in CVS.
Regarding PWLIB, I have intentionally added PWLIB in the opensipstack
source tree to avoid version conflicts. As much as possible, I would
try to keep the pwlib src updated everytime Craig Southeren releases new
stable version in voxgratia.
Joegen
Leandro Caetano Gonçalves Lustosa wrote:
> Hi,
>
> I'm using Visual C++ 2005 to compile OSS and I'm having some problems.
>
> I was trying to build OSS and I got an error on vblasterlid.cxx.
> Something like "undefined symbol i".
>
> I fixed it with the following modifications:
>
> // for (int i = 3; ringOn && i > 0; i--) { ---- original line
> int i; // my line
> for ( i = 3; ringOn && i > 0; i--) { // my line
>
> vBlaster.WriteCommand(VoipBlasterInterface::Command_VOL_3);
> vBlaster.WriteCommand(VoipBlasterInterface::Command_RING_OFF);
> vBlaster.WriteCommand(VoipBlasterInterface::Command_PHONE_ON);
>
> PTimer timer(2000);
>
> do {
> status = vBlaster.ReadStatus(STARTUP_TIMEOUT);
> if (status != VoipBlasterInterface::Status_Empty)
> HandleStatus(status);
> } while (ringOn && timer.IsRunning());
> }
>
> if (i == 0) { //
> this line could not be build
> PTRACE(3, "vBlaster\tCould not initialise");
> return FALSE;
> }
>
> -----------
> Did anybody have the same problem?
> Any advice?
>
> Other doubt...
>
> Do I need to use the pwlib from
> "http://www.voxgratia.org/downloads.html" or use the pwlib that is
> inside OSS?
>
>
> Thanks,
>
>
|
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From: Joegen E. B. <joe...@gm...> - 2006-11-07 23:13:16
|
Hi Laurent, 'make bothnoshared' should work. Joegen laurent schweizer wrote: > Hi all, > > I have downloaded openSBC 1.1.1 and when i try to compile it, I have > this error > > Laurent > > [root@byll4 opensbc]# make all > make[1]: Entering directory `/home/opensbc' > set -e; for i in /home/opensipstack /home/opensipstack; do make -C $i > debugdepend debug; done > make[2]: Entering directory `/home/opensipstack' > Created dependencies. > set -e; make -C src/pwlib debugdepend; make -C src debugdepend; > make[3]: Entering directory `/home/opensipstack/src/pwlib' > Created dependencies. > set -e; make -C src/ptlib/unix debugdepend; make -C plugins debugdepend; > make[4]: Entering directory `/home/opensipstack/src/pwlib/src/ptlib/unix' > Created dependencies. > make[4]: Leaving directory `/home/opensipstack/src/pwlib/src/ptlib/unix' > make[4]: Entering directory `/home/opensipstack/src/pwlib/plugins' > Created dependencies. > set -e; > make[4]: Leaving directory `/home/opensipstack/src/pwlib/plugins' > make[3]: Leaving directory `/home/opensipstack/src/pwlib' > make[3]: Entering directory `/home/opensipstack/src' > make[3]: *** No rule to make target `uicmp.cxx', needed by > `/home/opensipstack/lib/obj_linux_x86_64_d/uicmp.dep'. Stop. > make[3]: Leaving directory `/home/opensipstack/src' > make[2]: *** [debugdepend] Error 2 > make[2]: Leaving directory `/home/opensipstack' > make[1]: *** [libs] Error 2 > make[1]: Leaving directory `/home/opensbc' > make: *** [debuglibs] Error 2 > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Using Tomcat but need to do more? Need to support web services, security? > Get stuff done quickly with pre-integrated technology to make your job easier > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From:
<lca...@gm...> - 2006-11-07 23:06:44
|
Hi,
I'm using Visual C++ 2005 to compile OSS and I'm having some problems.
I was trying to build OSS and I got an error on vblasterlid.cxx.
Something like "undefined symbol i".
I fixed it with the following modifications:
// for (int i = 3; ringOn && i > 0; i--) { ---- original line
int i; // my line
for ( i = 3; ringOn && i > 0; i--) { // my line
vBlaster.WriteCommand(VoipBlasterInterface::Command_VOL_3);
vBlaster.WriteCommand(VoipBlasterInterface::Command_RING_OFF);
vBlaster.WriteCommand(VoipBlasterInterface::Command_PHONE_ON);
PTimer timer(2000);
do {
status = vBlaster.ReadStatus(STARTUP_TIMEOUT);
if (status != VoipBlasterInterface::Status_Empty)
HandleStatus(status);
} while (ringOn && timer.IsRunning());
}
if (i == 0) { //
this line could not be build
PTRACE(3, "vBlaster\tCould not initialise");
return FALSE;
}
-----------
Did anybody have the same problem?
Any advice?
Other doubt...
Do I need to use the pwlib from
"http://www.voxgratia.org/downloads.html" or use the pwlib that is
inside OSS?
Thanks,
--
Leandro
|
|
From: laurent s. <lau...@gm...> - 2006-11-07 18:09:28
|
Hi all, I have downloaded openSBC 1.1.1 and when i try to compile it, I have this error Laurent [root@byll4 opensbc]# make all make[1]: Entering directory `/home/opensbc' set -e; for i in /home/opensipstack /home/opensipstack; do make -C $i debugdepend debug; done make[2]: Entering directory `/home/opensipstack' Created dependencies. set -e; make -C src/pwlib debugdepend; make -C src debugdepend; make[3]: Entering directory `/home/opensipstack/src/pwlib' Created dependencies. set -e; make -C src/ptlib/unix debugdepend; make -C plugins debugdepend; make[4]: Entering directory `/home/opensipstack/src/pwlib/src/ptlib/unix' Created dependencies. make[4]: Leaving directory `/home/opensipstack/src/pwlib/src/ptlib/unix' make[4]: Entering directory `/home/opensipstack/src/pwlib/plugins' Created dependencies. set -e; make[4]: Leaving directory `/home/opensipstack/src/pwlib/plugins' make[3]: Leaving directory `/home/opensipstack/src/pwlib' make[3]: Entering directory `/home/opensipstack/src' make[3]: *** No rule to make target `uicmp.cxx', needed by `/home/opensipstack/lib/obj_linux_x86_64_d/uicmp.dep'. Stop. make[3]: Leaving directory `/home/opensipstack/src' make[2]: *** [debugdepend] Error 2 make[2]: Leaving directory `/home/opensipstack' make[1]: *** [libs] Error 2 make[1]: Leaving directory `/home/opensbc' make: *** [debuglibs] Error 2 |
|
From: Joegen E. B. <joe...@gm...> - 2006-11-07 08:59:55
|
Resend... Joegen E. Baclor wrote: > Leandro, > > I've been contemplating on SRTP support for quite a while now and > there seems to be libsrtp *(srtp*.*sourceforge*.net/*srtp*.html) which > opensipstack could use for the SRTP stack. But the first thing that > needs to be done is Multipart Mime support for content-type to be able > to offer both secure and none secure SDP in a single INVITE. I'll > keep the list posted as soon as something new comes up regarding this > subject. I am also choosing the best key exchange mechanism to be > used. Haven't decided if S/MIME is appropriate or would it be > Mikey. I am open to suggestions from the list. > > Joegen > > > Leandro Caetano Gonçalves Lustosa wrote: >> Hi Joegen, >> >> Thank you for the answer. >> >> Is SRTP a goal for the OSS development? >> >> Leandro >> >> ---------- Forwarded message ---------- >> From: Joegen E. Baclor <jo...@pl...> >> Date: Nov 2, 2006 5:09 PM >> Subject: Re: [OpenSIPStack] SRTP >> To: ope...@li..., lca...@gm... >> >> >> Leandro Caetano Gonçalves Lustosa wrote: >> >>> Hi, >>> >>> Reading the features information on OSS website I noticed something >>> regarded to "SIP/RTP Hash Encryption". Is it a SRTP and SIP over TLS >>> implementation? >>> >>> Thanks, >>> >>> >> Hi Leandro, >> >> Unfortunately not. The Encryption mentioned in the web site is a none >> standard simple encryption supported by Grand Stream and Inter Edge and >> its not SRTP. >> >> Joegen >> >> >> > > |
|
From: Joegen E. B. <joe...@gm...> - 2006-11-07 05:31:04
|
Leandro, I've been contemplating on SRTP support for quite a while now and there seems to be libsrtp *(srtp*.*sourceforge*.net/*srtp*.html) which opensipstack could use for the SRTP stack. But the first thing that needs to be done is Multipart Mime support for content-type to be able to offer both secure and none secure SDP in a single INVITE. I'll keep the list posted as soon as something new comes up regarding this subject. I am also choosing the best key exchange mechanism to be used. Haven't decided if S/MIME is appropriate or would it be Mikey. I am open to suggestions from the list. Joegen Leandro Caetano Gonçalves Lustosa wrote: > Hi Joegen, > > Thank you for the answer. > > Is SRTP a goal for the OSS development? > > Leandro > > ---------- Forwarded message ---------- > From: Joegen E. Baclor <jo...@pl...> > Date: Nov 2, 2006 5:09 PM > Subject: Re: [OpenSIPStack] SRTP > To: ope...@li..., lca...@gm... > > > Leandro Caetano Gonçalves Lustosa wrote: > >> Hi, >> >> Reading the features information on OSS website I noticed something >> regarded to "SIP/RTP Hash Encryption". Is it a SRTP and SIP over TLS >> implementation? >> >> Thanks, >> >> > Hi Leandro, > > Unfortunately not. The Encryption mentioned in the web site is a none > standard simple encryption supported by Grand Stream and Inter Edge and > its not SRTP. > > Joegen > > > |
|
From:
<lca...@gm...> - 2006-11-06 11:59:03
|
Hi Joegen, Thank you for the answer. Is SRTP a goal for the OSS development? Leandro ---------- Forwarded message ---------- From: Joegen E. Baclor <jo...@pl...> Date: Nov 2, 2006 5:09 PM Subject: Re: [OpenSIPStack] SRTP To: ope...@li..., lca...@gm... Leandro Caetano Gon=E7alves Lustosa wrote: > Hi, > > Reading the features information on OSS website I noticed something > regarded to "SIP/RTP Hash Encryption". Is it a SRTP and SIP over TLS > implementation? > > Thanks, > Hi Leandro, Unfortunately not. The Encryption mentioned in the web site is a none standard simple encryption supported by Grand Stream and Inter Edge and its not SRTP. Joegen --=20 Leandro |
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