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From: Andre M. <an...@ma...> - 2008-10-27 15:38:07
|
Hi Joegen, my tests against the latest CVS version were successfull. Calls are properly terminated now - we do not see ghost connections anymore. Many thanks for the help and the quick fix. Andre jo...@op... schrieb: > Andre, > > If it's not much trouble. Please provide level 5 log and wireshark > capture from the sbc box. > > Joegen > > André Mamitzsch wrote: >> Hello Joegen, >> >> I have tested against the latest CVS version, 1.1.5-27, still no >> success. The softswitch still responds with a "400 Bad Request". >> >> Regards, >> >> Andre >> >> jo...@op... schrieb: >> >>> Andre, >>> >>> This is a known problem area in OpenSBC. Since we are a B2BUA and not a >>> proxy we terminate our dialogs based on successful processing of a BYE >>> request. OpenSBC does not wait for a final response before it destroys >>> connections. This ensures that the SBC does not end up with ghost >>> sessions as well if a certain UA is not able to handle authenticated BYE >>> requests. This is the second time I have received this report regarding >>> this problem. Can you provide me a with a level 5 log for this >>> problematic call offlist? I'll see if an easy workaround is possible. >>> If you could, please also attach a wire shark capture of the successful >>> call you made using a UA directly calling your softswitch. My e-mail >>> address is jo...@op... >>> >>> Joegen >>> >>> André Mamitzsch wrote: >>> >>>> Hello, >>>> >>>> I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is >>>> as follows: >>>> >>>> The OpenSBC is configured in B2BUAUpperReg mode. >>>> >>>> I place a call from a SIP client (A) to a PSTN Phone (B). The connection >>>> is established whithout any problems. A goes on hook and sends a BYE to >>>> which the SBC immediately replies with a 200 OK. >>>> >>>> Another BYE message is being sent out to the softswitch platform. >>>> >>>> The softswitch challenges authentication and therefore responds with a >>>> 401 unauthorized message. Since the SBC, at least that's my assumption, >>>> has already successfully terminated the internal connection there is no >>>> match and the requested authenticated BYE will never be send out. This >>>> behavior results in ghost connections and the billing runs until B goes >>>> on-hook as well. >>>> >>>> I have tested the same scenario with my SIP client directly connected to >>>> our softswitch platform and did not detect the problem there. Upon >>>> receiving the 401 unauthorized message, the client resends the BYE >>>> containing the authentication information. >>>> >>>> I have checked the RFC 3261 this morning and tried to find out whether >>>> it is a regular behavior to reply to a BYE message with authentication >>>> challenge or not. I could not find any hint there. >>>> >>>> How difficult is it to change this behavior and wait for confirmation >>>> before terminating the entire connection ? >>>> >>>> Regards, >>>> >>>> Andre >>>> >>>> >>>> ------------------------------------------------------------------------- >>>> 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >>>> >>>> >>>> >>> ------------------------------------------------------------------------- >>> 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 >>> >> >> ------------------------------------------------------------------------- >> 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: 8.0.175 / Virus Database: 270.8.3/1748 - Release Date: 10/26/2008 7:53 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 |
From: <jo...@op...> - 2008-10-27 13:53:09
|
Andre, If it's not much trouble. Please provide level 5 log and wireshark capture from the sbc box. Joegen André Mamitzsch wrote: > Hello Joegen, > > I have tested against the latest CVS version, 1.1.5-27, still no > success. The softswitch still responds with a "400 Bad Request". > > Regards, > > Andre > > jo...@op... schrieb: > >> Andre, >> >> This is a known problem area in OpenSBC. Since we are a B2BUA and not a >> proxy we terminate our dialogs based on successful processing of a BYE >> request. OpenSBC does not wait for a final response before it destroys >> connections. This ensures that the SBC does not end up with ghost >> sessions as well if a certain UA is not able to handle authenticated BYE >> requests. This is the second time I have received this report regarding >> this problem. Can you provide me a with a level 5 log for this >> problematic call offlist? I'll see if an easy workaround is possible. >> If you could, please also attach a wire shark capture of the successful >> call you made using a UA directly calling your softswitch. My e-mail >> address is jo...@op... >> >> Joegen >> >> André Mamitzsch wrote: >> >>> Hello, >>> >>> I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is >>> as follows: >>> >>> The OpenSBC is configured in B2BUAUpperReg mode. >>> >>> I place a call from a SIP client (A) to a PSTN Phone (B). The connection >>> is established whithout any problems. A goes on hook and sends a BYE to >>> which the SBC immediately replies with a 200 OK. >>> >>> Another BYE message is being sent out to the softswitch platform. >>> >>> The softswitch challenges authentication and therefore responds with a >>> 401 unauthorized message. Since the SBC, at least that's my assumption, >>> has already successfully terminated the internal connection there is no >>> match and the requested authenticated BYE will never be send out. This >>> behavior results in ghost connections and the billing runs until B goes >>> on-hook as well. >>> >>> I have tested the same scenario with my SIP client directly connected to >>> our softswitch platform and did not detect the problem there. Upon >>> receiving the 401 unauthorized message, the client resends the BYE >>> containing the authentication information. >>> >>> I have checked the RFC 3261 this morning and tried to find out whether >>> it is a regular behavior to reply to a BYE message with authentication >>> challenge or not. I could not find any hint there. >>> >>> How difficult is it to change this behavior and wait for confirmation >>> before terminating the entire connection ? >>> >>> Regards, >>> >>> Andre >>> >>> >>> ------------------------------------------------------------------------- >>> 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >>> >>> >>> >> >> ------------------------------------------------------------------------- >> 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 >> > > > ------------------------------------------------------------------------- > 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: 8.0.175 / Virus Database: 270.8.3/1748 - Release Date: 10/26/2008 7:53 PM > > |
From: André M. <an...@ma...> - 2008-10-27 13:40:43
|
Hello Joegen, I have tested against the latest CVS version, 1.1.5-27, still no success. The softswitch still responds with a "400 Bad Request". Regards, Andre jo...@op... schrieb: > Andre, > > This is a known problem area in OpenSBC. Since we are a B2BUA and not a > proxy we terminate our dialogs based on successful processing of a BYE > request. OpenSBC does not wait for a final response before it destroys > connections. This ensures that the SBC does not end up with ghost > sessions as well if a certain UA is not able to handle authenticated BYE > requests. This is the second time I have received this report regarding > this problem. Can you provide me a with a level 5 log for this > problematic call offlist? I'll see if an easy workaround is possible. > If you could, please also attach a wire shark capture of the successful > call you made using a UA directly calling your softswitch. My e-mail > address is jo...@op... > > Joegen > > André Mamitzsch wrote: >> Hello, >> >> I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is >> as follows: >> >> The OpenSBC is configured in B2BUAUpperReg mode. >> >> I place a call from a SIP client (A) to a PSTN Phone (B). The connection >> is established whithout any problems. A goes on hook and sends a BYE to >> which the SBC immediately replies with a 200 OK. >> >> Another BYE message is being sent out to the softswitch platform. >> >> The softswitch challenges authentication and therefore responds with a >> 401 unauthorized message. Since the SBC, at least that's my assumption, >> has already successfully terminated the internal connection there is no >> match and the requested authenticated BYE will never be send out. This >> behavior results in ghost connections and the billing runs until B goes >> on-hook as well. >> >> I have tested the same scenario with my SIP client directly connected to >> our softswitch platform and did not detect the problem there. Upon >> receiving the 401 unauthorized message, the client resends the BYE >> containing the authentication information. >> >> I have checked the RFC 3261 this morning and tried to find out whether >> it is a regular behavior to reply to a BYE message with authentication >> challenge or not. I could not find any hint there. >> >> How difficult is it to change this behavior and wait for confirmation >> before terminating the entire connection ? >> >> Regards, >> >> Andre >> >> >> ------------------------------------------------------------------------- >> 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >> >> > > > > ------------------------------------------------------------------------- > 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 |
From: <jo...@op...> - 2008-10-27 12:42:40
|
Hi André, It turned out that merely resending the authorization header from the first INVITE is not enough if the UAC and the UAS both supports qop="auth". Please try the new patch in CVS and confirm if your server and OpenSBC can now properly inter-operate. Joegen André Mamitzsch wrote: > Hello Joegen, > > I have compiled the latest version from CVS. There is a small typo in > opensbc/SBCB2BUAEndPoint.h which must be corrected to get it working. > > --- SBCB2BUAEndPoint.h.orig 2008-10-27 12:15:46.000000000 +0100 > +++ SBCB2BUAEndPoint.h 2008-10-27 12:03:28.000000000 +0100 > @@ -50,7 +50,7 @@ > #ifndef SBCB2BUAENDPOINT_H > #define SBCB2BUAENDPOINT_H > > -#include "B2BUAendPoint.h" > +#include "B2BUAEndPoint.h" > > namespace B2BUA > { > > I have tested the call scenario again, but it still fails. The > softswitch is responding with a "400 Bad Request" to the BYE message > now. I have sent you the wireshark offline. > > Regards, > > Andre > > > jo...@op... schrieb: > >> Andre, >> >> This is a known problem area in OpenSBC. Since we are a B2BUA and not a >> proxy we terminate our dialogs based on successful processing of a BYE >> request. OpenSBC does not wait for a final response before it destroys >> connections. This ensures that the SBC does not end up with ghost >> sessions as well if a certain UA is not able to handle authenticated BYE >> requests. This is the second time I have received this report regarding >> this problem. Can you provide me a with a level 5 log for this >> problematic call offlist? I'll see if an easy workaround is possible. >> If you could, please also attach a wire shark capture of the successful >> call you made using a UA directly calling your softswitch. My e-mail >> address is jo...@op... >> >> Joegen >> >> André Mamitzsch wrote: >> >>> Hello, >>> >>> I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is >>> as follows: >>> >>> The OpenSBC is configured in B2BUAUpperReg mode. >>> >>> I place a call from a SIP client (A) to a PSTN Phone (B). The connection >>> is established whithout any problems. A goes on hook and sends a BYE to >>> which the SBC immediately replies with a 200 OK. >>> >>> Another BYE message is being sent out to the softswitch platform. >>> >>> The softswitch challenges authentication and therefore responds with a >>> 401 unauthorized message. Since the SBC, at least that's my assumption, >>> has already successfully terminated the internal connection there is no >>> match and the requested authenticated BYE will never be send out. This >>> behavior results in ghost connections and the billing runs until B goes >>> on-hook as well. >>> >>> I have tested the same scenario with my SIP client directly connected to >>> our softswitch platform and did not detect the problem there. Upon >>> receiving the 401 unauthorized message, the client resends the BYE >>> containing the authentication information. >>> >>> I have checked the RFC 3261 this morning and tried to find out whether >>> it is a regular behavior to reply to a BYE message with authentication >>> challenge or not. I could not find any hint there. >>> >>> How difficult is it to change this behavior and wait for confirmation >>> before terminating the entire connection ? >>> >>> Regards, >>> >>> Andre >>> >>> >>> ------------------------------------------------------------------------- >>> 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >>> >>> >>> >> >> ------------------------------------------------------------------------- >> 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 >> > > > ------------------------------------------------------------------------- > 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: 8.0.175 / Virus Database: 270.8.3/1748 - Release Date: 10/26/2008 7:53 PM > > |
From: <jo...@op...> - 2008-10-27 12:08:42
|
Ok I see the fault. Thanks for the wireshark capture. Fix will be in shortly. André Mamitzsch wrote: > Hello Joegen, > > I have compiled the latest version from CVS. There is a small typo in > opensbc/SBCB2BUAEndPoint.h which must be corrected to get it working. > > --- SBCB2BUAEndPoint.h.orig 2008-10-27 12:15:46.000000000 +0100 > +++ SBCB2BUAEndPoint.h 2008-10-27 12:03:28.000000000 +0100 > @@ -50,7 +50,7 @@ > #ifndef SBCB2BUAENDPOINT_H > #define SBCB2BUAENDPOINT_H > > -#include "B2BUAendPoint.h" > +#include "B2BUAEndPoint.h" > > namespace B2BUA > { > > I have tested the call scenario again, but it still fails. The > softswitch is responding with a "400 Bad Request" to the BYE message > now. I have sent you the wireshark offline. > > Regards, > > Andre > > > jo...@op... schrieb: >> Andre, >> >> This is a known problem area in OpenSBC. Since we are a B2BUA and >> not a proxy we terminate our dialogs based on successful processing >> of a BYE request. OpenSBC does not wait for a final response before >> it destroys connections. This ensures that the SBC does not end up >> with ghost sessions as well if a certain UA is not able to handle >> authenticated BYE requests. This is the second time I have received >> this report regarding this problem. Can you provide me a with a >> level 5 log for this problematic call offlist? I'll see if an easy >> workaround is possible. If you could, please also attach a wire >> shark capture of the successful call you made using a UA directly >> calling your softswitch. My e-mail address is jo...@op... >> >> Joegen >> >> André Mamitzsch wrote: >>> Hello, >>> >>> I have seen a strange behavior with the OpenSBC1.1.5RC1. The >>> scenario is as follows: >>> >>> The OpenSBC is configured in B2BUAUpperReg mode. >>> >>> I place a call from a SIP client (A) to a PSTN Phone (B). The >>> connection is established whithout any problems. A goes on hook and >>> sends a BYE to which the SBC immediately replies with a 200 OK. >>> >>> Another BYE message is being sent out to the softswitch platform. >>> >>> The softswitch challenges authentication and therefore responds with >>> a 401 unauthorized message. Since the SBC, at least that's my >>> assumption, has already successfully terminated the internal >>> connection there is no match and the requested authenticated BYE >>> will never be send out. This behavior results in ghost connections >>> and the billing runs until B goes on-hook as well. >>> >>> I have tested the same scenario with my SIP client directly >>> connected to our softswitch platform and did not detect the problem >>> there. Upon receiving the 401 unauthorized message, the client >>> resends the BYE containing the authentication information. >>> >>> I have checked the RFC 3261 this morning and tried to find out >>> whether it is a regular behavior to reply to a BYE message with >>> authentication challenge or not. I could not find any hint there. >>> >>> How difficult is it to change this behavior and wait for >>> confirmation before terminating the entire connection ? >>> >>> Regards, >>> >>> Andre >>> >>> >>> ------------------------------------------------------------------------- >>> >>> 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: 8.0.175 / Virus >>> Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >>> >>> >> >> >> >> ------------------------------------------------------------------------- >> >> 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: 8.0.175 / Virus Database: 270.8.3/1748 - Release Date: 10/26/2008 7:53 PM > > |
From: André M. <an...@ma...> - 2008-10-27 11:29:32
|
Hello Joegen, I have compiled the latest version from CVS. There is a small typo in opensbc/SBCB2BUAEndPoint.h which must be corrected to get it working. --- SBCB2BUAEndPoint.h.orig 2008-10-27 12:15:46.000000000 +0100 +++ SBCB2BUAEndPoint.h 2008-10-27 12:03:28.000000000 +0100 @@ -50,7 +50,7 @@ #ifndef SBCB2BUAENDPOINT_H #define SBCB2BUAENDPOINT_H -#include "B2BUAendPoint.h" +#include "B2BUAEndPoint.h" namespace B2BUA { I have tested the call scenario again, but it still fails. The softswitch is responding with a "400 Bad Request" to the BYE message now. I have sent you the wireshark offline. Regards, Andre jo...@op... schrieb: > Andre, > > This is a known problem area in OpenSBC. Since we are a B2BUA and not a > proxy we terminate our dialogs based on successful processing of a BYE > request. OpenSBC does not wait for a final response before it destroys > connections. This ensures that the SBC does not end up with ghost > sessions as well if a certain UA is not able to handle authenticated BYE > requests. This is the second time I have received this report regarding > this problem. Can you provide me a with a level 5 log for this > problematic call offlist? I'll see if an easy workaround is possible. > If you could, please also attach a wire shark capture of the successful > call you made using a UA directly calling your softswitch. My e-mail > address is jo...@op... > > Joegen > > André Mamitzsch wrote: >> Hello, >> >> I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is >> as follows: >> >> The OpenSBC is configured in B2BUAUpperReg mode. >> >> I place a call from a SIP client (A) to a PSTN Phone (B). The connection >> is established whithout any problems. A goes on hook and sends a BYE to >> which the SBC immediately replies with a 200 OK. >> >> Another BYE message is being sent out to the softswitch platform. >> >> The softswitch challenges authentication and therefore responds with a >> 401 unauthorized message. Since the SBC, at least that's my assumption, >> has already successfully terminated the internal connection there is no >> match and the requested authenticated BYE will never be send out. This >> behavior results in ghost connections and the billing runs until B goes >> on-hook as well. >> >> I have tested the same scenario with my SIP client directly connected to >> our softswitch platform and did not detect the problem there. Upon >> receiving the 401 unauthorized message, the client resends the BYE >> containing the authentication information. >> >> I have checked the RFC 3261 this morning and tried to find out whether >> it is a regular behavior to reply to a BYE message with authentication >> challenge or not. I could not find any hint there. >> >> How difficult is it to change this behavior and wait for confirmation >> before terminating the entire connection ? >> >> Regards, >> >> Andre >> >> >> ------------------------------------------------------------------------- >> 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >> >> > > > > ------------------------------------------------------------------------- > 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 |
From: <jo...@op...> - 2008-10-27 10:26:49
|
André, Please try patches in CVS and update the mailing list about the result of your test. I think by simply resending the original Authorization header sent in the initial INVITE doe subsequent in-dialog requests should do the trick in your case. Thanks. Joegen André Mamitzsch wrote: > Hello Joegen, > > many thanks for your help in respect to this topic. Please find the > requested traces attached. The archive contains the following files: > > 1. diagram.txt - ladder logic digram inc. all IP Addresses > 2. b2bua-2008-10-27-24959.log - Level 5 trace from OpenSBC > 3. trace_20081027_sbc_sip.cap - Wireshark trace from OpenSBC for the > call in question > 4. trace_20081027_bts10200.cap - Wireshark trace for call with UA > directly connected to BTS10200 > > If you need anything else, please let me know. > Andre, > > This is a known problem area in OpenSBC. Since we are a B2BUA and not a > proxy we terminate our dialogs based on successful processing of a BYE > request. OpenSBC does not wait for a final response before it destroys > connections. This ensures that the SBC does not end up with ghost > sessions as well if a certain UA is not able to handle authenticated BYE > requests. This is the second time I have received this report regarding > this problem. Can you provide me a with a level 5 log for this > problematic call offlist? I'll see if an easy workaround is possible. > If you could, please also attach a wire shark capture of the successful > call you made using a UA directly calling your softswitch. My e-mail > address is jo...@op... > > Joegen > > André Mamitzsch wrote: > >> Hello, >> >> I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is >> as follows: >> >> The OpenSBC is configured in B2BUAUpperReg mode. >> >> I place a call from a SIP client (A) to a PSTN Phone (B). The connection >> is established whithout any problems. A goes on hook and sends a BYE to >> which the SBC immediately replies with a 200 OK. >> >> Another BYE message is being sent out to the softswitch platform. >> >> The softswitch challenges authentication and therefore responds with a >> 401 unauthorized message. Since the SBC, at least that's my assumption, >> has already successfully terminated the internal connection there is no >> match and the requested authenticated BYE will never be send out. This >> behavior results in ghost connections and the billing runs until B goes >> on-hook as well. >> >> I have tested the same scenario with my SIP client directly connected to >> our softswitch platform and did not detect the problem there. Upon >> receiving the 401 unauthorized message, the client resends the BYE >> containing the authentication information. >> >> I have checked the RFC 3261 this morning and tried to find out whether >> it is a regular behavior to reply to a BYE message with authentication >> challenge or not. I could not find any hint there. >> >> How difficult is it to change this behavior and wait for confirmation >> before terminating the entire connection ? >> >> Regards, >> >> Andre >> >> >> ------------------------------------------------------------------------- >> 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM >> >> >> > > > > ------------------------------------------------------------------------- > 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: 8.0.175 / Virus Database: 270.8.3/1746 - Release Date: 10/25/2008 5:55 PM > > |
From: <jo...@op...> - 2008-10-26 01:58:53
|
Hi Voice, I have already responded to the thread you have opened in the sipx mailing list. This is better discussed there. I am not a (direct) user of sipx myself so I would rather bring this discussion there so that people who do know sipx first hand can comment. Joegen voice wrote: > Hi Joegen > > I'ts been over 14 months now. Have you guys completed testing the > Pingtel/sipX PBX and removing the SIP Trunk registration requiement? I > believe your sip trunk feature was forcing registration via > username/password, which cause some SIP Trunk provider to choke on. > > r > > > ----- Original Message ----- > From: "Joegen E. Baclor" <joe...@gm...> > To: <ope...@li...> > Sent: Wednesday, August 22, 2007 8:52 PM > Subject: Re: [OpenSIPStack] sipXecs and openSBC > > > >> voice wrote: >> >>> Hi >>> >>> I have setup a sipXecs box and a standalone openSBC box. >>> >>> i want to use the sipX 'SIP Trunk' Gateway to define and point to >>> > openSBC from sipX. > >>> My goal is to use sipX to register USERS and PHONES and to use openSBC >>> > to pass SiP traffic to remote SBC, which will handle DID < == > PSTN > >>> ATA <==> sipX <==> openSBC <==> SBC/PSTN >>> >>> Can this be done with openSBC in the middle? >>> >>> How do i set up openSBC to do 'SIP egress and ingress TRUNKing'? >>> >>> >> If you are pertaining to the SIP Trunking being discussed in the sipX >> user list, then this functionality is not yet complete. However, >> OpenSBC can already be used for basic trunking needs like traversing NAT >> and routing to your carrier as a B2BUA. If your carrier needs to do a >> separate authentication, then you must wait till I announce the >> trunking feature completion. This will happen soon so watch this space. >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > > > ------------------------------------------------------------------------- > 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: 8.0.175 / Virus Database: 270.8.3/1746 - Release Date: 10/25/2008 5:55 PM > > |
From: voice <vo...@ne...> - 2008-10-25 18:32:04
|
Hi Joegen I'ts been over 14 months now. Have you guys completed testing the Pingtel/sipX PBX and removing the SIP Trunk registration requiement? I believe your sip trunk feature was forcing registration via username/password, which cause some SIP Trunk provider to choke on. r ----- Original Message ----- From: "Joegen E. Baclor" <joe...@gm...> To: <ope...@li...> Sent: Wednesday, August 22, 2007 8:52 PM Subject: Re: [OpenSIPStack] sipXecs and openSBC > voice wrote: > > Hi > > > > I have setup a sipXecs box and a standalone openSBC box. > > > > i want to use the sipX 'SIP Trunk' Gateway to define and point to openSBC from sipX. > > > > My goal is to use sipX to register USERS and PHONES and to use openSBC to pass SiP traffic to remote SBC, which will handle DID < == > PSTN > > > > ATA <==> sipX <==> openSBC <==> SBC/PSTN > > > > Can this be done with openSBC in the middle? > > > > How do i set up openSBC to do 'SIP egress and ingress TRUNKing'? > > > > > If you are pertaining to the SIP Trunking being discussed in the sipX > user list, then this functionality is not yet complete. However, > OpenSBC can already be used for basic trunking needs like traversing NAT > and routing to your carrier as a B2BUA. If your carrier needs to do a > separate authentication, then you must wait till I announce the > trunking feature completion. This will happen soon so watch this space. > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: <jo...@op...> - 2008-10-25 14:47:30
|
Andre, This is a known problem area in OpenSBC. Since we are a B2BUA and not a proxy we terminate our dialogs based on successful processing of a BYE request. OpenSBC does not wait for a final response before it destroys connections. This ensures that the SBC does not end up with ghost sessions as well if a certain UA is not able to handle authenticated BYE requests. This is the second time I have received this report regarding this problem. Can you provide me a with a level 5 log for this problematic call offlist? I'll see if an easy workaround is possible. If you could, please also attach a wire shark capture of the successful call you made using a UA directly calling your softswitch. My e-mail address is jo...@op... Joegen André Mamitzsch wrote: > Hello, > > I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is > as follows: > > The OpenSBC is configured in B2BUAUpperReg mode. > > I place a call from a SIP client (A) to a PSTN Phone (B). The connection > is established whithout any problems. A goes on hook and sends a BYE to > which the SBC immediately replies with a 200 OK. > > Another BYE message is being sent out to the softswitch platform. > > The softswitch challenges authentication and therefore responds with a > 401 unauthorized message. Since the SBC, at least that's my assumption, > has already successfully terminated the internal connection there is no > match and the requested authenticated BYE will never be send out. This > behavior results in ghost connections and the billing runs until B goes > on-hook as well. > > I have tested the same scenario with my SIP client directly connected to > our softswitch platform and did not detect the problem there. Upon > receiving the 401 unauthorized message, the client resends the BYE > containing the authentication information. > > I have checked the RFC 3261 this morning and tried to find out whether > it is a regular behavior to reply to a BYE message with authentication > challenge or not. I could not find any hint there. > > How difficult is it to change this behavior and wait for confirmation > before terminating the entire connection ? > > Regards, > > Andre > > > ------------------------------------------------------------------------- > 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: 8.0.175 / Virus Database: 270.8.2/1743 - Release Date: 10/24/2008 8:33 AM > > |
From: André M. <an...@ma...> - 2008-10-24 14:18:33
|
Hello, I have seen a strange behavior with the OpenSBC1.1.5RC1. The scenario is as follows: The OpenSBC is configured in B2BUAUpperReg mode. I place a call from a SIP client (A) to a PSTN Phone (B). The connection is established whithout any problems. A goes on hook and sends a BYE to which the SBC immediately replies with a 200 OK. Another BYE message is being sent out to the softswitch platform. The softswitch challenges authentication and therefore responds with a 401 unauthorized message. Since the SBC, at least that's my assumption, has already successfully terminated the internal connection there is no match and the requested authenticated BYE will never be send out. This behavior results in ghost connections and the billing runs until B goes on-hook as well. I have tested the same scenario with my SIP client directly connected to our softswitch platform and did not detect the problem there. Upon receiving the 401 unauthorized message, the client resends the BYE containing the authentication information. I have checked the RFC 3261 this morning and tried to find out whether it is a regular behavior to reply to a BYE message with authentication challenge or not. I could not find any hint there. How difficult is it to change this behavior and wait for confirmation before terminating the entire connection ? Regards, Andre |
From: <jo...@op...> - 2008-10-24 04:18:13
|
Hi Marshall, I am creating a new thread for this. Since there is lack of documentation currently, I am providing you the next best thing. I coded a template user agent and demonstrated how to do outbound calls. Attached is the cxx file. I have compiled and tested this code to be working using the current opensipstack in CVS. HTH, Joegen > For outgoing calls, we need something like this: > > 1. Our app will setup an RTP ip/port to listen on (UDP). > > 2. Our app will give this RTP ip/port information (as well as any > other necessary info) to the opensipstack > to make a call (INVITE and such). > > 3. The SIP stack will negotiate with the destination SIP stack > (wherever that might be). The negotiation, > of course, will tell us what ip/port to send the RTP for outgoing > voice. We need a mechanism to receive > this info after the SIP portion of the call as been accepted by the > foreign site. > > 4. Now the SIP part is done (w/o using the opensipstack RTP > handling facilities) > > 5. Now it is the job of our code to send/receive RTP from the > foreign host, and walaa all it happy. > > 6. We also need to be able to send/receive the call terminations > (BYEs) and such. > > > > For incoming calls, we need something like this: > > 1. We setup the SIP stack to listen on 5060. > > 2. It receives incoming calls, (potentially gives this to our code > for approval). > Or maybe we just want to allow the SIP stack to handle everything SIP > relate > > 3. The SIP stack will need to know what RTP port we want to listen > on (maybe it should ask us) > > 4. Once a call is setup, again, it is our code that will be > sending/receiving RTP. > > > |
From: Marshall J. <MJ...@ef...> - 2008-10-23 18:08:47
|
This is what we want. We have an RTP stack of our own that we want to use. We need an interface to a SIP stack to initiate and receive SIP call sessions. We do NOT want the part of opensipstack that does the RTP. For outgoing calls, we need something like this: 1. Our app will setup an RTP ip/port to listen on (UDP). 2. Our app will give this RTP ip/port information (as well as any other necessary info) to the opensipstack to make a call (INVITE and such). 3. The SIP stack will negotiate with the destination SIP stack (wherever that might be). The negotiation, of course, will tell us what ip/port to send the RTP for outgoing voice. We need a mechanism to receive this info after the SIP portion of the call as been accepted by the foreign site. 4. Now the SIP part is done (w/o using the opensipstack RTP handling facilities) 5. Now it is the job of our code to send/receive RTP from the foreign host, and walaa all it happy. 6. We also need to be able to send/receive the call terminations (BYEs) and such. For incoming calls, we need something like this: 1. We setup the SIP stack to listen on 5060. 2. It receives incoming calls, (potentially gives this to our code for approval). Or maybe we just want to allow the SIP stack to handle everything SIP relate 3. The SIP stack will need to know what RTP port we want to listen on (maybe it should ask us) 4. Once a call is setup, again, it is our code that will be sending/receiving RTP. -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Thursday, October 23, 2008 10:41 AM To: ope...@li... Subject: Re: [OpenSIPStack] TCP and TLS support Documentation is scarse for opensipstack, at least as of the moment. Please give more info on what type of application you would want to build so people can point you to the right direction. For starters, CallSession and CallSessionManager subclass is the way to go for UA/Softphone implementation. Marshall Jobe wrote: > Yes... > > I want to access the SIP interface of opensipstack and let our code > handle the RTP and voice (codecs and all). > > I just want to use the SIP portion. > > I have found no technical documentation whatsoever and don't know where > to go from here. > > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, October 22, 2008 8:49 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] TCP and TLS support > > Marshall Jobe wrote: > >> Help needed... >> This is what we need to do. >> >> We have our own RTP stack that we want to use. We just want to use the >> SIP >> portion of the opensipstack and have the SIP portion tell us when to >> > set > >> up >> an RTP call. We want to manage the RTP portion completely. >> >> How do we do this using opensipstack? >> >> > > Please clarify. You want to use your own RTP stack. Does this include > using your own codecs and audio input output steams as well? > > > > > ------------------------------------------------------------------------ > - > 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 > > > > ------------------------------------------------------------------------ - > 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: 8.0.175 / Virus Database: 270.8.2/1740 - Release Date: 10/22/2008 7:24 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 |
From: <jo...@op...> - 2008-10-23 15:41:31
|
Documentation is scarse for opensipstack, at least as of the moment. Please give more info on what type of application you would want to build so people can point you to the right direction. For starters, CallSession and CallSessionManager subclass is the way to go for UA/Softphone implementation. Marshall Jobe wrote: > Yes... > > I want to access the SIP interface of opensipstack and let our code > handle the RTP and voice (codecs and all). > > I just want to use the SIP portion. > > I have found no technical documentation whatsoever and don't know where > to go from here. > > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, October 22, 2008 8:49 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] TCP and TLS support > > Marshall Jobe wrote: > >> Help needed... >> This is what we need to do. >> >> We have our own RTP stack that we want to use. We just want to use the >> SIP >> portion of the opensipstack and have the SIP portion tell us when to >> > set > >> up >> an RTP call. We want to manage the RTP portion completely. >> >> How do we do this using opensipstack? >> >> > > Please clarify. You want to use your own RTP stack. Does this include > using your own codecs and audio input output steams as well? > > > > > ------------------------------------------------------------------------ > - > 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 > > > > ------------------------------------------------------------------------- > 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: 8.0.175 / Virus Database: 270.8.2/1740 - Release Date: 10/22/2008 7:24 PM > > |
From: <jo...@op...> - 2008-10-23 15:36:41
|
This means windns.h was not located by configure.exe. DNS API is usually part of the platform SDK that goes with visual studio. This is a dependency of opensipstack in windows builds. Please confirm if you have windns.h and dnsapi.lib in your system. If not, install the platform SDK for you MSVC compiler Marshall Jobe wrote: > > > > SIPSrvRecord.cxx will not compile... > > > > .\src\SIPSrvRecord.cxx(452) : error C2653: 'PDNS' : is not a class or > namespace name > > .\src\SIPSrvRecord.cxx(452) : error C2065: 'SRVRecordList' : > undeclared identifier > > .\src\SIPSrvRecord.cxx(452) : error C2065: 'srvUdpRecords' : > undeclared identifier > > .\src\SIPSrvRecord.cxx(453) : error C2653: 'PDNS' : is not a class or > namespace name > > .\src\SIPSrvRecord.cxx(453) : error C2065: 'SRVRecordList' : > undeclared identifier > > .\src\SIPSrvRecord.cxx(453) : error C2065: 'srvTcpRecords' : > undeclared identifier > > .\src\SIPSrvRecord.cxx(464) : error C2065: 'srvUdpRecords' : > undeclared identifier > > > > ... > > > > > > I have tracked it down. The opensipstack/include/ptclib/pdns.h file is > #if'd out. > > > > At the top it says #if P_DNS since this is not defined in the project, > it doesn't build. > > > > Why is this not defined? I'm using your .sln file. OpenSIPStack.sln. > > I am using > > > > > > > > -----Original Message----- > From: jo...@op... [mailto:joe...@gm...] > Sent: Wednesday, October 22, 2008 8:49 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] TCP and TLS support > > > > Marshall Jobe wrote: > > > Help needed... > > > This is what we need to do. > > > > > > We have our own RTP stack that we want to use. We just want to use the > > > SIP > > > portion of the opensipstack and have the SIP portion tell us when to set > > > up > > > an RTP call. We want to manage the RTP portion completely. > > > > > > How do we do this using opensipstack? > > > > > > > Please clarify. You want to use your own RTP stack. Does this include > > using your own codecs and audio input output steams as well? > > > > > > > > > > ------------------------------------------------------------------------- > > 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: 8.0.175 / Virus Database: 270.8.2/1740 - Release Date: 10/22/2008 7:24 PM > > |
From: Marshall J. <MJ...@ef...> - 2008-10-23 14:55:44
|
Yes... I want to access the SIP interface of opensipstack and let our code handle the RTP and voice (codecs and all). I just want to use the SIP portion. I have found no technical documentation whatsoever and don't know where to go from here. -----Original Message----- From: jo...@op... [mailto:joe...@gm...] Sent: Wednesday, October 22, 2008 8:49 PM To: ope...@li... Subject: Re: [OpenSIPStack] TCP and TLS support Marshall Jobe wrote: > Help needed... > This is what we need to do. > > We have our own RTP stack that we want to use. We just want to use the > SIP > portion of the opensipstack and have the SIP portion tell us when to set > up > an RTP call. We want to manage the RTP portion completely. > > How do we do this using opensipstack? > Please clarify. You want to use your own RTP stack. Does this include using your own codecs and audio input output steams as well? ------------------------------------------------------------------------ - 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 |
From: <jo...@op...> - 2008-10-23 02:42:00
|
Hi Everyone, I am proud to announce that we have published the first release candidate for OpenSBC 1.1.5. Source bundle and Windows installer for this release can be found in the download section of opensipstack.org ( http://www.opensipstack.org/downloads.html ) and the open source sip forum site ( http://www.opensourcesip.org:8080/clearspacex/docs/DOC-1001 ). Enjoy, Joegen |
From: <jo...@op...> - 2008-10-23 01:49:11
|
Marshall Jobe wrote: > Help needed... > This is what we need to do. > > We have our own RTP stack that we want to use. We just want to use the > SIP > portion of the opensipstack and have the SIP portion tell us when to set > up > an RTP call. We want to manage the RTP portion completely. > > How do we do this using opensipstack? > Please clarify. You want to use your own RTP stack. Does this include using your own codecs and audio input output steams as well? |
From: Marshall J. <MJ...@ef...> - 2008-10-22 15:46:30
|
Help needed... This is what we need to do. We have our own RTP stack that we want to use. We just want to use the SIP portion of the opensipstack and have the SIP portion tell us when to set up an RTP call. We want to manage the RTP portion completely. How do we do this using opensipstack? |
From: OpenSIPStack F. <ope...@op...> - 2008-10-22 06:03:58
|
> > Is TCP and TLS transport protocols are supported by OpenSipStack right now? I am sure on TLS is not but still can anybody confirm this? > Only UDP is supported in OpenSIPStack current. TCP support is incomplete and TLS is, of course, dependent on TCP. Joegen |
From: OpenSIPStack F. <ope...@op...> - 2008-10-22 05:58:39
|
Hi there, Is TCP and TLS transport protocols are supported by OpenSipStack right now? I am sure on TLS is not but still can anybody confirm this? Thanks in advance. Regards, Jaydeep |
From: Andre M. <an...@ma...> - 2008-10-20 16:02:51
|
Hello Joegen, many thanks for the reply - I read it and I immediately found the problem. I missed the ";from=sipphone.com" statement in the route configuration - it's working now. Many thanks again ! Andre jo...@op... schrieb: > Andre, > > Sorry for the late response. I fired up an instance opensbc just a > while ago. Added [sip:*;from=sipphone.com] sip:proxy01.sipphone.com and > the from rewrite worked. So I am not sure what you are reporting here. > I can't reproduce it. I have attached the log for my test call. You > will see the the call went in with > From: joegen <sip:77...@th...>;tag=fc663e5c > and went out as > From: joegen <sip:77...@si...>;tag=fc663e5c > > > Joegen > > Andre Mamitzsch wrote: >> Hello Joegen, >> >> did you find some time to back-track the changes ? An update on that >> would be highly appreciated. >> >> Regards, >> >> Andre >> >> Joegen E. Baclor schrieb: >> >>> Andre >>> >>> I will try to back-track the changes. That revision was checked in >>> by another developer. I need to confer with him before I conuld >>> confirm what really changed since then. >>> Joegen >>> >>> André Mamitzsch wrote: >>> >>>> -----BEGIN PGP SIGNED MESSAGE----- >>>> Hash: SHA1 >>>> >>>> Hello Joegen, >>>> >>>> the rewrite of the domain in the from field seems to work until >>>> 1.1.5-10. After that, and I compiled almost everything what I could get >>>> from CVS, it is not working anymore. >>>> >>>> I tried to have a look at the differences in order to find out why it >>>> changed and what was changed. I could identify that in Version 1.28 of >>>> the Router.cxx the Feature: >>>> >>>> Router.cxx: * Added feature to rewrite the FROM domain to a specific >>>> domain in the B2BUA routes >>>> >>>> was added. >>>> >>>> At the moment I' m unable to find a good starting point - if you could >>>> give me a small hint please... >>>> >>>> Andre >>>> >>>> >>>> Andre Mamitzsch schrieb: >>>> >>>>> Hello Joegen, >>>>> >>>>> thanks for your reply. I'm with you regarding the 50/50 - I checked >>>>> the RFC3261 this morning. >>>>> >>>>> > Never the less, I'll find time to make the domain rewrite for from >>>>> > configurable via B2BUA Route as well. >>>>> >>>>> Please wait. I found something confusing today which I cannot >>>>> explain yet. I disabled the "Rewrite-Request-URI" and >>>>> "Rewrite-TO-URI" and things start working. The from field value ist >>>>> replaced with the target domain (abc.net) while the to: field >>>>> remains unchanged and contains the local domain value (xyz.net). >>>>> >>>>> I' m totally confused now - I need to overthink that and come back >>>>> to you as soon as I verified the scenario. >>>>> >>>>> Regards, >>>>> >>>>> Andre >>>>> >>>>> >>>>> jo...@op... schrieb: >>>>> >>>>>> Andre, >>>>>> >>>>>> I am 50/50 on this one. Half of the argument is as you have >>>>>> already stated in this post. The other half is, this is the first >>>>>> time I have encountered a GW that filters INVITE based on the from >>>>>> header. From is a very weak form of identity assertion since it >>>>>> can be forged very easily using any softphone. This is not >>>>>> mentioning that RFC 3261 not only allows From to be forged easily >>>>>> but also allows it to be anonymized. For example >>>>>> sip:ano...@an...valid to indicate that you want your >>>>>> identity to be private is aperfectly valid from header. Thus, an >>>>>> implementation that relies on From to assert identity is >>>>>> tantamount to a multitude of interoperability issues. >>>>>> >>>>>> Never the less, I'll find time to make the domain rewrite for from >>>>>> configurable via B2BUA Route as well. I guess these sorts of >>>>>> issues is the main reason why SBC's are invented anyway. >>>>>> >>>>>> Joegen >>>>>> >>>>>> Andre Mamitzsch wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I just need some clarification regarding the domain rewriting >>>>>>> concept of the OpenSBC. >>>>>>> >>>>>>> We are using the SBC in upper registration mode. The upper >>>>>>> registration works fine, all register messages are "hijacked" and >>>>>>> domain rewriting is performed. So, everything is fine here. >>>>>>> >>>>>>> >>>>>>> If I try to place a call, I receive the message "forbidden AOR" >>>>>>> since no rewriting in the From: field is performed while the >>>>>>> domain in the To: is replaced as expected. >>>>>>> >>>>>>> In my opinion, the local domain (xyz.net) should be replaced by >>>>>>> the target domain (abc.net) in the From: as well. Or am I wrong >>>>>>> here ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> Andre >>>>>>> >>>>>>> Our setup: >>>>>>> >>>>>>> >>>>>>> +--------+ +-------+ +---------+ >>>>>>> + SIP UA +------------------+ oSBC +-------------------+ SIP Net + >>>>>>> +--------+ +-------+ +---------+ >>>>>>> 192.168.10.3 10.70.3.200 10.70.5.85 >>>>>>> 12...@si... sip.abc.net >>>>>>> >>>>>>> >>>>>>> 2008/10/01 22:35:05.384 DTL: [CID=0x0e3e] ICT(3121173204) >>>>>>> Event(SIPMessage) - SIP/2.0 403 Forbidden AOR >>>>>>> >>>>>>> 2008/10/01 22:35:05.384 DBG: [CID=0x0e3e] TRANSACTION: (ICT) >>>>>>> SIP/2.0 403 >>>>>>> Forbidden AOR State: 3 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 INF: [CID=0x0e3e] >>> ACK >>>>>>> sip:78...@si... >>>>>>> SIP/2.0 DST: 10.70.5.85:5060:UDP SRC: 10.70.3.200:5060 enc=0 >>>>>>> bytes=656 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] ACK >>>>>>> sip:78...@si... SIP/2.0 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] From: "123456" >>>>>>> <sip:12...@si...>;tag=ed94ab1c >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] To: "789123" >>>>>>> <sip:78...@si...>;tag=1_1146_t160890_14h5 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Via: SIP/2.0/UDP >>>>>>> 10.70.3.200:5060;iid=22771;branch=z9hG4bK4ec38706668edd119c5bdb9c33a35a27;uas-addr=10.70.5.85;rport >>>>>>> >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] CSeq: 1 ACK >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Call-ID: >>>>>>> ZTMzZjllOTQ1M2UwZGI0YWY2OGRlNjJjNTE2MGZmOTc.-0x0004 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Contact: >>>>>>> <sip:123456@10.70.3.200:5060> >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] User-Agent: OpenSBC >>>>>>> v1.1.5-13 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Max-Forwards: 70 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Allow: INVITE, ACK, >>>>>>> CANCEL, >>>>>>> OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Supported: timer >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Session-Expires: 1800 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Min-SE: 90 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Content-Length: 0 >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>>>> >>>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>>>> >>>>>>> ------------------------------------------------------------------------- >>>>>>> >>>>>>> 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: 8.0.173 / Virus >>>>>>> Database: 270.7.5/1702 - Release Date: 10/1/2008 9:05 AM >>>>>>> >>>>>>> >>>>>> ------------------------------------------------------------------------- >>>>>> >>>>>> 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 >>>>>> >>>>> ------------------------------------------------------------------------- >>>>> >>>>> 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 >>>>> >>>> -----BEGIN PGP SIGNATURE----- >>>> Version: GnuPG v1.4.7 (MingW32) >>>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org >>>> >>>> iD8DBQFI61SIQKZIuVrbkWoRAmYkAJ499q3C8J94fe6kOO3nM0fVr46IiwCfcnW+ >>>> cRvR/59pqaqnzhqiilg0B84= >>>> =WEQ/ >>>> -----END PGP SIGNATURE----- >>>> >>>> ------------------------------------------------------------------------- >>>> >>>> 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: 8.0.173 / Virus >>>> Database: 270.7.6/1711 - Release Date: 10/6/2008 5:37 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 >>> >> >> ------------------------------------------------------------------------- >> 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: 8.0.173 / Virus Database: >> 270.8.1/1733 - Release Date: 10/19/2008 6:02 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 |
From: Robert V. <ro...@dc...> - 2008-10-20 14:56:15
|
Hi, >From a little testing, it would seem that ATLSIP has a memory leak. I am more than willing to accept that it might be me with the leak, but I can't find any in the simple app I wrote. Can anyone maby confirm this for me? Regards, Robert |
From: <jo...@op...> - 2008-10-20 10:24:33
|
Andre, Sorry for the late response. I fired up an instance opensbc just a while ago. Added [sip:*;from=sipphone.com] sip:proxy01.sipphone.com and the from rewrite worked. So I am not sure what you are reporting here. I can't reproduce it. I have attached the log for my test call. You will see the the call went in with From: joegen <sip:77...@th...>;tag=fc663e5c and went out as From: joegen <sip:77...@si...>;tag=fc663e5c Joegen Andre Mamitzsch wrote: > Hello Joegen, > > did you find some time to back-track the changes ? An update on that > would be highly appreciated. > > Regards, > > Andre > > Joegen E. Baclor schrieb: > >> Andre >> >> I will try to back-track the changes. That revision was checked in by >> another developer. I need to confer with him before I conuld confirm >> what really changed since then. >> >> Joegen >> >> André Mamitzsch wrote: >> >>> -----BEGIN PGP SIGNED MESSAGE----- >>> Hash: SHA1 >>> >>> Hello Joegen, >>> >>> the rewrite of the domain in the from field seems to work until >>> 1.1.5-10. After that, and I compiled almost everything what I could get >>> from CVS, it is not working anymore. >>> >>> I tried to have a look at the differences in order to find out why it >>> changed and what was changed. I could identify that in Version 1.28 of >>> the Router.cxx the Feature: >>> >>> Router.cxx: * Added feature to rewrite the FROM domain to a specific >>> domain in the B2BUA routes >>> >>> was added. >>> >>> At the moment I' m unable to find a good starting point - if you could >>> give me a small hint please... >>> >>> Andre >>> >>> >>> Andre Mamitzsch schrieb: >>> >>> >>>> Hello Joegen, >>>> >>>> thanks for your reply. I'm with you regarding the 50/50 - I checked the >>>> RFC3261 this morning. >>>> >>>> > Never the less, I'll find time to make the domain rewrite for from >>>> > configurable via B2BUA Route as well. >>>> >>>> Please wait. I found something confusing today which I cannot explain >>>> yet. I disabled the "Rewrite-Request-URI" and "Rewrite-TO-URI" and >>>> things start working. The from field value ist replaced with the target >>>> domain (abc.net) while the to: field remains unchanged and contains the >>>> local domain value (xyz.net). >>>> >>>> I' m totally confused now - I need to overthink that and come back to >>>> you as soon as I verified the scenario. >>>> >>>> Regards, >>>> >>>> Andre >>>> >>>> >>>> jo...@op... schrieb: >>>> >>>> >>>>> Andre, >>>>> >>>>> I am 50/50 on this one. Half of the argument is as you have already >>>>> stated in this post. The other half is, this is the first time I have >>>>> encountered a GW that filters INVITE based on the from header. From is >>>>> a very weak form of identity assertion since it can be forged very >>>>> easily using any softphone. This is not mentioning that RFC 3261 not >>>>> only allows From to be forged easily but also allows it to be >>>>> anonymized. For example sip:ano...@an...valid to indicate >>>>> that you want your identity to be private is aperfectly valid from >>>>> header. Thus, an implementation that relies on From to assert identity >>>>> is tantamount to a multitude of interoperability issues. >>>>> >>>>> Never the less, I'll find time to make the domain rewrite for from >>>>> configurable via B2BUA Route as well. I guess these sorts of issues is >>>>> the main reason why SBC's are invented anyway. >>>>> >>>>> Joegen >>>>> >>>>> Andre Mamitzsch wrote: >>>>> >>>>> >>>>>> Hi, >>>>>> >>>>>> I just need some clarification regarding the domain rewriting concept of >>>>>> the OpenSBC. >>>>>> >>>>>> We are using the SBC in upper registration mode. The upper registration >>>>>> works fine, all register messages are "hijacked" and domain rewriting is >>>>>> performed. So, everything is fine here. >>>>>> >>>>>> >>>>>> If I try to place a call, I receive the message "forbidden AOR" since no >>>>>> rewriting in the From: field is performed while the domain in the To: >>>>>> is replaced as expected. >>>>>> >>>>>> In my opinion, the local domain (xyz.net) should be replaced by the >>>>>> target domain (abc.net) in the From: as well. Or am I wrong here ? >>>>>> >>>>>> Regards, >>>>>> >>>>>> Andre >>>>>> >>>>>> Our setup: >>>>>> >>>>>> >>>>>> +--------+ +-------+ +---------+ >>>>>> + SIP UA +------------------+ oSBC +-------------------+ SIP Net + >>>>>> +--------+ +-------+ +---------+ >>>>>> 192.168.10.3 10.70.3.200 10.70.5.85 >>>>>> 12...@si... sip.abc.net >>>>>> >>>>>> >>>>>> 2008/10/01 22:35:05.384 DTL: [CID=0x0e3e] ICT(3121173204) >>>>>> Event(SIPMessage) - SIP/2.0 403 Forbidden AOR >>>>>> >>>>>> 2008/10/01 22:35:05.384 DBG: [CID=0x0e3e] TRANSACTION: (ICT) SIP/2.0 403 >>>>>> Forbidden AOR State: 3 >>>>>> >>>>>> 2008/10/01 22:35:05.385 INF: [CID=0x0e3e] >>> ACK sip:78...@si... >>>>>> SIP/2.0 DST: 10.70.5.85:5060:UDP SRC: 10.70.3.200:5060 enc=0 bytes=656 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] ACK sip:78...@si... SIP/2.0 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] From: "123456" >>>>>> <sip:12...@si...>;tag=ed94ab1c >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] To: "789123" >>>>>> <sip:78...@si...>;tag=1_1146_t160890_14h5 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Via: SIP/2.0/UDP >>>>>> 10.70.3.200:5060;iid=22771;branch=z9hG4bK4ec38706668edd119c5bdb9c33a35a27;uas-addr=10.70.5.85;rport >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] CSeq: 1 ACK >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Call-ID: >>>>>> ZTMzZjllOTQ1M2UwZGI0YWY2OGRlNjJjNTE2MGZmOTc.-0x0004 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Contact: >>>>>> <sip:123456@10.70.3.200:5060> >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] User-Agent: OpenSBC v1.1.5-13 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Max-Forwards: 70 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Allow: INVITE, ACK, CANCEL, >>>>>> OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Supported: timer >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Session-Expires: 1800 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Min-SE: 90 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Content-Length: 0 >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>>> >>>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>>> >>>>>> ------------------------------------------------------------------------- >>>>>> 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: 8.0.173 / Virus Database: 270.7.5/1702 - Release Date: 10/1/2008 9:05 AM >>>>>> >>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------- >>>>> 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 >>>>> >>>>> >>>> ------------------------------------------------------------------------- >>>> 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 >>>> >>>> >>> -----BEGIN PGP SIGNATURE----- >>> Version: GnuPG v1.4.7 (MingW32) >>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org >>> >>> iD8DBQFI61SIQKZIuVrbkWoRAmYkAJ499q3C8J94fe6kOO3nM0fVr46IiwCfcnW+ >>> cRvR/59pqaqnzhqiilg0B84= >>> =WEQ/ >>> -----END PGP SIGNATURE----- >>> >>> ------------------------------------------------------------------------- >>> 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: 8.0.173 / Virus Database: 270.7.6/1711 - Release Date: 10/6/2008 5:37 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 >> > > ------------------------------------------------------------------------- > 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: 8.0.173 / Virus Database: 270.8.1/1733 - Release Date: 10/19/2008 6:02 PM > > |
From: Andre M. <an...@ma...> - 2008-10-18 17:04:43
|
Hello Joegen, did you find some time to back-track the changes ? An update on that would be highly appreciated. Regards, Andre Joegen E. Baclor schrieb: > Andre > > I will try to back-track the changes. That revision was checked in by > another developer. I need to confer with him before I conuld confirm > what really changed since then. > > Joegen > > André Mamitzsch wrote: >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hello Joegen, >> >> the rewrite of the domain in the from field seems to work until >> 1.1.5-10. After that, and I compiled almost everything what I could get >> from CVS, it is not working anymore. >> >> I tried to have a look at the differences in order to find out why it >> changed and what was changed. I could identify that in Version 1.28 of >> the Router.cxx the Feature: >> >> Router.cxx: * Added feature to rewrite the FROM domain to a specific >> domain in the B2BUA routes >> >> was added. >> >> At the moment I' m unable to find a good starting point - if you could >> give me a small hint please... >> >> Andre >> >> >> Andre Mamitzsch schrieb: >> >>> Hello Joegen, >>> >>> thanks for your reply. I'm with you regarding the 50/50 - I checked the >>> RFC3261 this morning. >>> >>> > Never the less, I'll find time to make the domain rewrite for from >>> > configurable via B2BUA Route as well. >>> >>> Please wait. I found something confusing today which I cannot explain >>> yet. I disabled the "Rewrite-Request-URI" and "Rewrite-TO-URI" and >>> things start working. The from field value ist replaced with the target >>> domain (abc.net) while the to: field remains unchanged and contains the >>> local domain value (xyz.net). >>> >>> I' m totally confused now - I need to overthink that and come back to >>> you as soon as I verified the scenario. >>> >>> Regards, >>> >>> Andre >>> >>> >>> jo...@op... schrieb: >>> >>>> Andre, >>>> >>>> I am 50/50 on this one. Half of the argument is as you have already >>>> stated in this post. The other half is, this is the first time I have >>>> encountered a GW that filters INVITE based on the from header. From is >>>> a very weak form of identity assertion since it can be forged very >>>> easily using any softphone. This is not mentioning that RFC 3261 not >>>> only allows From to be forged easily but also allows it to be >>>> anonymized. For example sip:ano...@an...valid to indicate >>>> that you want your identity to be private is aperfectly valid from >>>> header. Thus, an implementation that relies on From to assert identity >>>> is tantamount to a multitude of interoperability issues. >>>> >>>> Never the less, I'll find time to make the domain rewrite for from >>>> configurable via B2BUA Route as well. I guess these sorts of issues is >>>> the main reason why SBC's are invented anyway. >>>> >>>> Joegen >>>> >>>> Andre Mamitzsch wrote: >>>> >>>>> Hi, >>>>> >>>>> I just need some clarification regarding the domain rewriting concept of >>>>> the OpenSBC. >>>>> >>>>> We are using the SBC in upper registration mode. The upper registration >>>>> works fine, all register messages are "hijacked" and domain rewriting is >>>>> performed. So, everything is fine here. >>>>> >>>>> >>>>> If I try to place a call, I receive the message "forbidden AOR" since no >>>>> rewriting in the From: field is performed while the domain in the To: >>>>> is replaced as expected. >>>>> >>>>> In my opinion, the local domain (xyz.net) should be replaced by the >>>>> target domain (abc.net) in the From: as well. Or am I wrong here ? >>>>> >>>>> Regards, >>>>> >>>>> Andre >>>>> >>>>> Our setup: >>>>> >>>>> >>>>> +--------+ +-------+ +---------+ >>>>> + SIP UA +------------------+ oSBC +-------------------+ SIP Net + >>>>> +--------+ +-------+ +---------+ >>>>> 192.168.10.3 10.70.3.200 10.70.5.85 >>>>> 12...@si... sip.abc.net >>>>> >>>>> >>>>> 2008/10/01 22:35:05.384 DTL: [CID=0x0e3e] ICT(3121173204) >>>>> Event(SIPMessage) - SIP/2.0 403 Forbidden AOR >>>>> >>>>> 2008/10/01 22:35:05.384 DBG: [CID=0x0e3e] TRANSACTION: (ICT) SIP/2.0 403 >>>>> Forbidden AOR State: 3 >>>>> >>>>> 2008/10/01 22:35:05.385 INF: [CID=0x0e3e] >>> ACK sip:78...@si... >>>>> SIP/2.0 DST: 10.70.5.85:5060:UDP SRC: 10.70.3.200:5060 enc=0 bytes=656 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] ACK sip:78...@si... SIP/2.0 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] From: "123456" >>>>> <sip:12...@si...>;tag=ed94ab1c >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] To: "789123" >>>>> <sip:78...@si...>;tag=1_1146_t160890_14h5 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Via: SIP/2.0/UDP >>>>> 10.70.3.200:5060;iid=22771;branch=z9hG4bK4ec38706668edd119c5bdb9c33a35a27;uas-addr=10.70.5.85;rport >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] CSeq: 1 ACK >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Call-ID: >>>>> ZTMzZjllOTQ1M2UwZGI0YWY2OGRlNjJjNTE2MGZmOTc.-0x0004 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Contact: >>>>> <sip:123456@10.70.3.200:5060> >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] User-Agent: OpenSBC v1.1.5-13 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Max-Forwards: 70 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Allow: INVITE, ACK, CANCEL, >>>>> OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Supported: timer >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Session-Expires: 1800 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Min-SE: 90 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] Content-Length: 0 >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>> >>>>> 2008/10/01 22:35:05.385 DBG: [CID=0x0e3e] >>>>> >>>>> ------------------------------------------------------------------------- >>>>> 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 - 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