[Sip] SIP GW Register
"FCG Huang Qiang \(B300\)" <Qiang.F.Huang@alcatel-sbell.com.cn> Fri, 24 September 2004 10:54 UTC
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Date: Fri, 24 Sep 2004 18:47:42 +0800
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Thread-Topic: Sip Digest, Vol 5, Issue 17
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From: "FCG Huang Qiang (B300)" <Qiang.F.Huang@alcatel-sbell.com.cn>
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Subject: [Sip] SIP GW Register
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Question 1: Accordint to RFC 3261 10.2.2: Use of the "*" Contact header field value allows a registering UA to remove all bindings associated with an address-of-record without knowing their precise values. So if we use contact * and expire time zero can remove all contact bindings for the user, is it or not? Question 2: For the registrar discovery mechanism in the RFC 3261 charpter 10.2.6: we cand send register message with Request-URI "sip.mcast.net" to find registrar, and how the registrar reply with it? how we can get the registrar information, sucha as domain,ip address and port from the registrar response ? Question 3: Not like In H323 GK register mechanism the field termination type can be used to distinguish the GW or termiantion. In SIP GW can the telephone number prefix be used in the from and to field to send to registrar for telephone number routing? if ok, maybe are there several user to register with registrar for one GW. if not How SIP GW can register with registrar? -----ÔʼÓʼþ----- ·¢¼þÈË: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]´ú±í sip-request@ietf.org ·¢ËÍʱ¼ä: 2004Äê9ÔÂ23ÈÕ 0:45 ÊÕ¼þÈË: sip@ietf.org Ö÷Ìâ: Sip Digest, Vol 5, Issue 17 Send Sip mailing list submissions to sip@ietf.org To subscribe or unsubscribe via the World Wide Web, visit https://www1.ietf.org/mailman/listinfo/sip or, via email, send a message with subject or body 'help' to sip-request@ietf.org You can reach the person managing the list at sip-owner@ietf.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Sip digest..." Today's Topics: 1. RE: call stateful ? (Thomas Gal) 2. RE: Need guidance in call transfer - SIP Based (Umesh Sharma) 3. [Sip]: SIP and SAML (Goeman Stefan) 4. Performance testing tools for SIP entities (brijesh nair) 5. RE: Performance testing tools for SIP entities (Thomas Johansson G (HF/EAB)) 6. Re: Performance testing tools for SIP entities (Bj?rn Gr?nesj?) 7. Voice Mail (Anurag Kabra) 8. SIP and MIP addressing (Thibault Renier) 9. Re: [Sip]: SIP and SAML (Richard Shockey) ---------------------------------------------------------------------- Message: 1 Date: Tue, 21 Sep 2004 17:47:06 -0700 From: "Thomas Gal" <ThomasGal@LumenVox.com> Subject: RE: [Sip] call stateful ? To: "'vikram'" <vikramb@aftek.com>, "'sip-ietf'" <sip@ietf.org> Message-ID: <LV-SVR5P22aoCZjL7av00000660@lv-svr.lumenvox.com> Content-Type: text/plain; charset="us-ascii" That means a proxy tracks the transaction ID not the session ID. For example once the invite has completed a proxy may know nothing about the call that it connected while for a brief period of time it was "aware" (read maintaining state) that it was in the middle of an invite command. -Tom > -----Original Message----- > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On > Behalf Of vikram > Sent: Thursday, September 16, 2004 6:26 AM > To: sip-ietf > Subject: [Sip] call stateful ? > > hello all, > Could anybody pls explain following from rfc 3261 > > " *A call stateful proxy is always transaction stateful, > but the converse is not necessarily true"* > > > regards > > Vikram > > _______________________________________________ > Sip mailing list https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sipping@ietf.org for new developments on the application of sip > ------------------------------ Message: 2 Date: Wed, 22 Sep 2004 09:53:46 +0530 From: "Umesh Sharma" <umesh_sharma@persistent.co.in> Subject: RE: [Sip] Need guidance in call transfer - SIP Based To: <sip-bounces@ietf.org>, "sip" <sip@ietf.org> Message-ID: <HNECIFNGLLKHFHHLCABCCELOCCAA.umesh_sharma@persistent.co.in> Content-Type: text/plain; charset="iso-8859-1" pls see inline. -----Original Message----- From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of Anurag Kabra (by way of Anurag Kabra<anuragk@aftek.com>) (by way of Anurag Kabra <anuragk@aftek.com>) Sent: Monday, September 20, 2004 1:24 PM To: sip@ietf.org Subject: [Sip] Need guidance in call transfer - SIP Based I am currently working on VoIP, and I am located in India. I have few queries which are bothering me, could you suggest me something on them. Below are my queries. 1. Do we do retransmission for Refer Request(& Notify) if it doesnot reach transferee. >> timeout/retransmission mechanism for refer is same as for bye, non-invite transactions; when notify times out it is not retransmitted & correp. subscription should be removed. 2. In a unattended call transfer, when does the dialog end i.e. normally with a BYE or Notify(subscriptio-state="terminated"). 3. Does the dialog exists even after BYE or subscription-state header in Notify will terminate the dialog. 2, 3>> my understanding from rfc 3265 bye would not terminate the dialog if there is subscription on it, subscriber-state="terminated" destroys the subscription and the dialog if no other application state is associated with the dialog (more in 3.3.4 rfc 3265). For Call transfer refer goes with in existing dialog. 4. Is 202 Accepted sent after getting a REFER request, ignored by the transferor. What is the effect of the response 202 Accepted? >> it indicates that the subscription has been accepted and that authorization to proceed with transfer may or may not have been granted. Also, that a notify will be sent immediately (3.1.4.1). It also creates a new dialog & subscription if the initial refer was not sent on a pre-existing dialog otherwise it just creates the new subscription with the dialog. This is all for now. Hope to hear from you soon. Thanks in advance. Best Regards Anurag. _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip ------------------------------ Message: 3 Date: Wed, 22 Sep 2004 10:30:55 +0200 From: Goeman Stefan <Stefan.Goeman@siemens.com> Subject: [Sip]: SIP and SAML To: "'sip@ietf.org'" <sip@ietf.org> Message-ID: <57FD2C3A246F76438CA6FDAD8FE9F195692195@hrtades7.atea.be> Content-Type: text/plain; charset="us-ascii" Hello, I was wondering if somebody already reviewed the ID draft-tschofenig-sip-saml-00.txt. This draft explicitly mentions that its main goal is not to realize single sign-on (SSO) functionality as in HTTP. However, I wonder if it would not be a good idea to realize SSO functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. This procedure authenticates the end-user when sending an initial SIP Register request to the P-CSCF. It must be noted that this is an MNO-centric approach; the MNO owning both a wireless access network and an IMS domain. However, it is not unthinkable that in the future there will exist entities that will only operate an IMS domain. In this case, end-users will be authenticated by the (wireless) access network provider. Then, I believe it makes sense that the access network provider would provide SAML assertions to the IMS operator providing information on the authentication status of the end-user. How this should be done is unclear to me; I do have some initial ideas though. Anybody willing to comment? Thanks in advance! Greetings, Stefan. Stefan Goeman SIEMENS ICM/ICN Stefan.Goeman@siemens.com <mailto:Stefan.Goeman@siemens.atea.be> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www1.ietf.org/pipermail/sip/attachments/20040922/c26a946e/attachment.htm ------------------------------ Message: 4 Date: Wed, 22 Sep 2004 02:21:31 -0700 (PDT) From: brijesh nair <briju31@yahoo.com> Subject: [Sip] Performance testing tools for SIP entities To: sip@ietf.org Message-ID: <20040922092131.31780.qmail@web14224.mail.yahoo.com> Content-Type: text/plain; charset=us-ascii Hi, I am looking for good performace testing tools (free or otherwise) which can be used to test various SIP entities (e.g B2BUA, Proxy Servers etc.). Can anyone help on this?? Thanks, Brijesh _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ------------------------------ Message: 5 Date: Wed, 22 Sep 2004 11:30:47 +0200 From: "Thomas Johansson G (HF/EAB)" <thomas.g.johansson@ericsson.com> Subject: RE: [Sip] Performance testing tools for SIP entities To: "'brijesh nair'" <briju31@yahoo.com>, sip@ietf.org Message-ID: <633A2BAC69E8D411819A00508BCF883C0DA5F66D@esealnt452.al.sw.ericsson.se> Content-Type: text/plain; charset="iso-8859-1" Sipp from HP might fulfil your needs (freeware): http://sourceforge.net/projects/sipp Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (xml based) scenarios, dynamic adjustement of call-rate and a comprehensive set of real-time statistics. BRs /Thomas -----Original Message----- From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of brijesh nair Sent: den 22 september 2004 11:22 To: sip@ietf.org Subject: [Sip] Performance testing tools for SIP entities Hi, I am looking for good performace testing tools (free or otherwise) which can be used to test various SIP entities (e.g B2BUA, Proxy Servers etc.). Can anyone help on this?? Thanks, Brijesh _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip ------------------------------ Message: 6 Date: Wed, 22 Sep 2004 11:32:07 +0200 (CEST) From: Bj?rn Gr?nesj? <bg@ingate.com> Subject: Re: [Sip] Performance testing tools for SIP entities To: brijesh nair <briju31@yahoo.com> Cc: sip@ietf.org Message-ID: <Pine.LNX.4.61.0409221129380.16071@conar.ingate.se> Content-Type: text/plain; charset="iso-8859-1" Try: SIPp - sipp.sourceforge.net //Björn On Wed, 22 Sep 2004, brijesh nair wrote: > Hi, > I am looking for good performace testing tools (free > or otherwise) which can be used to test various SIP > entities (e.g B2BUA, Proxy Servers etc.). Can anyone > help on this?? > > Thanks, > Brijesh > > > > > > _______________________________ > Do you Yahoo!? > Declare Yourself - Register online to vote today! > http://vote.yahoo.com > > _______________________________________________ > Sip mailing list https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sipping@ietf.org for new developments on the application of sip > ------------------------------ Message: 7 Date: Wed, 22 Sep 2004 18:10:31 +0530 From: Anurag Kabra <anuragk@aftek.com> Subject: [Sip] Voice Mail To: sip@ietf.org Message-ID: <200409221810.31676.anuragk@aftek.com> Content-Type: text/plain; charset="us-ascii" Hello Everyone I need some links detailing Voice Mail Client implementation based on SIP. Thank You in advance. Best Regards Anurag. ------------------------------ Message: 8 Date: Wed, 22 Sep 2004 16:13:48 +0200 From: "Thibault Renier" <toubix@kom.auc.dk> Subject: [Sip] SIP and MIP addressing To: <sip@ietf.org> Message-ID: <FCEGIMIHPNAAAKBFICFAOEOFCEAA.toubix@kom.auc.dk> Content-Type: text/plain; charset="iso-8859-1" Hi all, I am currently working on integrating SIP in architecture using Mobile IP for mobility support. My question concerns the addressing scheme: At the SIP layer a SIP URI specifies the destination point ("To" field), going from SIP hop to SIP hop. At the IP layer, the destination address is usually the same as the current IP address (Care of Address in the MIP jargon -CoA) that corresponds to the URI used in the "To" (I guess that the association URI/current IP address of a user is made possible during the registration procedure). When MIP is used at the IP layer, for communications from the Correspondent Node (CN) to the Mobile Node (MN), the destination address is the MN's home address and the packets should be intercepted by the Home Agent (HA). 1. What is the route in a system combining MIP for mobility support and SIP for signalling flow? Do the packets go through all the SIP servers/hops before applying the MIP routing solutions? 2. What does happen because of the 'conflict' of addresses? (SIP resolves the URI with the CoA, while IP uses the MN's home address at first) Thanks for your help /Tibo -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 2004 bytes Desc: not available Url : http://www1.ietf.org/pipermail/sip/attachments/20040922/87c2eb8d/winmail.bin ------------------------------ Message: 9 Date: Wed, 22 Sep 2004 10:24:46 -0400 From: Richard Shockey <richard@shockey.us> Subject: Re: [Sip]: SIP and SAML To: Goeman Stefan <Stefan.Goeman@siemens.com>, "'sip@ietf.org'" <sip@ietf.org> Message-ID: <6.1.0.6.2.20040922102253.04e18210@joy.songbird.com> Content-Type: text/plain; charset="us-ascii" At 04:30 AM 9/22/2004, Goeman Stefan wrote: >Hello, > >I was wondering if somebody already reviewed the ID >draft-tschofenig-sip-saml-00.txt. > >This draft explicitly mentions that its main goal is not to realize single >sign-on (SSO) functionality as in HTTP. >However, I wonder if it would not be a good idea to realize SSO >functionality with SAML in SIP. Right now, 3GPP has defined IMS-AKA. This >procedure authenticates the end-user when sending an initial SIP Register >request to the P-CSCF. It must be noted that this is an MNO-centric >approach; the MNO owning both a wireless access network and an IMS domain. > >However, it is not unthinkable that in the future there will exist >entities that will only operate an IMS domain. In this case, end-users >will be authenticated by the (wireless) access network provider. Then, I >believe it makes sense that the access network provider would provide SAML >assertions to the IMS operator providing information on the authentication >status of the end-user. IMHO this is only one useful application of SAML to the SIP environment. I was personally disappointed this document did not recieve more attention in SanDiego I hope it will be given a broader hearing in DC. There are a variety of issues related to SIP security and Identity that SAML could solve. >How this should be done is unclear to me; I do have some initial ideas >though. > >Anybody willing to comment? > >Thanks in advance! > >Greetings, >Stefan. > >Stefan Goeman >SIEMENS ICM/ICN ><mailto:Stefan.Goeman@siemens.atea.be>Stefan.Goeman@siemens.com > >_______________________________________________ >Sip mailing list https://www1.ietf.org/mailman/listinfo/sip >This list is for NEW development of the core SIP Protocol >Use sip-implementors@cs.columbia.edu for questions on current sip >Use sipping@ietf.org for new developments on the application of sip >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Richard Shockey, Senior Manager, Strategic Technology Initiatives NeuStar Inc. 46000 Center Oak Plaza - Sterling, VA 20166 sip:rshockey(at)iptel.org sip:57141@fwd.pulver.com ENUM +87810-13313-31331 PSTN Office +1 571.434.5651 PSTN Mobile: +1 703.593.2683, Fax: +1 815.333.1237 <mailto:richard(at)shockey.us> or <mailto:richard.shockey(at)neustar.biz> <http://www.neustar.biz> ; <http://www.enum.org> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www1.ietf.org/pipermail/sip/attachments/20040922/cb454d7a/attachment.htm ------------------------------ _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip End of Sip Digest, Vol 5, Issue 17 ********************************** _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip
- [Sip] SIP GW Register FCG Huang Qiang (B300)