Re: [Sip-implementors] [Sip] SIPit 20 survey summary
Mark R. Lindsey wrote: It is much more reasonable to expect the service-provider/enterprise to implement the location conveyance. They'd add the location in their proxies/B2BUAs/ALGs. For example, an enterprise building ALG could add its location before sending the call to an SP. What if there is no service provider at all? SIP phones may perform emergency calls on there own - without the need of a service provider. Otherwise - if your service provider is down - you can't make emergency calls. regards klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
[Sip-implementors] Question about ETag in PUBLISH
Hi! On the openser mailing list we had some discussion about the Etag. If a client sends its first PUBLISH it gets an ETag from the EAC. This will be used in the second PUBLISH request. Does the ETag in the response to the second publish must be changed (A), or must it be the same (B)? (A) PUBLISH 200 OK SIP-ETag: abc PUBLISH 200 OK SIP-If-Match: abcSIP-ETag: def PUBLISH SIP-If-Match: def (B) PUBLISH 200 OK SIP-ETag: abc PUBLISH 200 OK SIP-If-Match: abcSIP-ETag: abc PUBLISH SIP-If-Match: abc If the ETag must be changed for each PUBLISH (even from the same client), why? What is the benefit over a constant etag? thanks klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
Re: [Sip-implementors] RFC3327 Path header - how many support it?
B, Nataraju wrote: Comments inline... Hi all, Is there anyone out there with experience of several proxies? And know of their list of features? Specifically, I was wondering is RFC3327 widely supported? To me, it seems that proxy support of this is quite important. Openser supports it since version 1.1.0 http://www.openser.org/docs/modules/1.1.x/path.html [ABN] AFAIK, this is mainly comes into use when IMS implementations. I am not aware of any plane SIP implementations which would need this feature implementation. It is a great feature also in not IMS setups as it allows the usage of stateless outbound proxies and load balancers, which forward the requests to downstream routing proxies and registrar. This helps when setting up big distributed systems. regards klaus -- Klaus Darilion nic.at ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
[Sip-implementors] how to signal network provided CLI vs. user provided CLI
Hi! I can't find any reference how to signal network provided vs. user provided CLI in SIP. I once used the Remote-Party-ID header for this. Using the privacy header allows to signal CLIP/CLIR, but AFAIK there is no support for network/user provided CLI. I further wonder how it is possible to signal multiple CLIs within SIP. E.g. in Austria in the PSTN there may be 2 CLIs in the ISDN-SETUP message - a user provided CLI and a network provided CLI. Any help is appreciated (references to the respective RFCs is fine) thanks klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
Re: [Sip-implementors] Question regarding Record-Route
Joegen E. Baclor wrote: Hi Manju, Thanks for the response. Yes I know that the proxy would be able to properly route the response. However being able to deliver the response is not really the concern of my question. In the case of a 200 OK response to an INVITE, the UAC needs to know the Record-Route to be able to properly contruct succeeding requests within the dialog. How would the UAC be able to construct the Route Set without the Record-Route in the response? Consider the scenario where there are several record_route headers present in the response, but the one inserted by your proxy is missing. How should the proxy know at which location the missing REcord-Route header should be inserted? I think the proxy has to rely that the UAS behaves correct. regards klaus Joegen Manjunath Warad wrote: Hi Joegen, Proxy shouldn't bother about the Record-Route in the response. No need to insert Record-Route in response. Response traverses in the network using Via header. Rgds, Manju -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joegen E. Baclor Sent: Friday, March 31, 2006 2:45 PM To: sip-implementors@cs.columbia.edu Subject: [Sip-implementors] Question regarding Record-Route Hi, If a proxy record routes and the response it gets does not contain the record-route, what should be the correct behavior of the proxy. Should it insert the record-route before relaying the response? Joegen ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
Re: [Sip-implementors] Need sip User agent implemention help
Hi Yogesh! First, you should ask yourself if you really want to program a new SIP UA although there are already lots of them. There are also a lot of open source clients which can be adopted to your needs. You can take a look at http://www.pernau.at/kd/voip/index.html were you will find links to SIP stacks, RTP stacks and SIP softphones. Choose one of these and start reading the docs of the respective stack. regards klaus yogesh bist wrote: Hi This is Yogender Singh from Chandigarh (India).I am working on a project in which i have to use SIP protocol . I am works on only c and c++ programming. I wana to implement only sip user agent which only throught call to the proxy server. there is lot of Q in my mind ,How agent send the inviteation and how call can be establish with RTP, Pls help me , How can i use sip user agent If any one have source code pls send me or any help pls send me in my mail. thankx yogender singh - New Yahoo! Messenger with Voice. Call regular phones from your PC and save big. ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
Re: [Sip-implementors] SIP dialer - open source
Ira Kadin wrote: Dear List, Could someone please suggest me the open sources for the SIP phone dialer ? I do not understand exactly what you mean, but you can find lots of open source VoIP software at: http://www.pernau.at/kd/voip/ regards klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
Re: [Sip-implementors] Xlite-soft phone
Looks like people are mixing things. xlite is able to have to proxy on a certain port, but xlite is not able to have the callee at a certain port. Thus, having the proxy at port 5050 and and dialing sip:[EMAIL PROTECTED]:4000 will send an INVITE to the proxy at port 5050, but the request line is INVITE sip:[EMAIL PROTECTED] instead of INVITE sip:[EMAIL PROTECTED]:4000 regards klaus [EMAIL PROTECTED] wrote: Hi Gururaj Do you mean its a seperate release? can you please tell me the release or version? can you please give me some more details about it? Also in settings i don't see any option by which i can set it to use TCP transport. I am using X-Lite release 1105d build on LINUX 9.0. Also i can configure the proxy port in X-lite so that i can specify which port proxy is present(other than 5060) and it is working for that, but if i want to send an INVITE with/without proxy to some port like 5063, it is not sending. I tried to specify in URI also like sip:[EMAIL PROTECTED]:5063 but it is sending to port 5060 only. Thank you +Basu -Original Message- From: GururajBT 70707 [mailto:[EMAIL PROTECTED] Sent: Thu 12/29/2005 1:10 PM To: Basavaraj Puttagangaiah (WT01 - Product Engineering Solutions) Subject: Re: [Sip-implementors] Xlite-soft phone Hi Basu, You have to check the hop implimentaion. There only the ip and port will be mentioned. Regards, Gururaj BT ** This email and its attachments contain confidential information from HUAWEI, which is intended only for the person or entity whose address is listed above. Any use of the information contained herein in any way (including, but not limited to, total or partial disclosure, reproduction, or dissemination) by persons other than the intended recipient(s) is prohibited. If you receive this e-mail in error, please notify the sender by phone or email immediately and delete it! - Original Message - From: [EMAIL PROTECTED] Date: Thursday, December 29, 2005 11:26 am Subject: [Sip-implementors] Xlite-soft phone Hi I am using Xlite softphone for testing my sip application. I need to send INVITE to some other port other than 5060. I have tried mentioning it in Request-URI, but still it sends request to 5060 only. Has anybody tried sending it to other port(except 5060), if it is possible how? Is there any other mailing list specifically for Xlite? Thank you +Basu The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
[Sip-implementors] TLS certificate question
Hi! I'm trying to figure out how to make a certificate for a SIP proxy. RFC3262, section 4.1 states: For NAPTR records with SIPS protocol fields, (if the server is using a site certificate), the domain name in the query and the domain name in the replacement field MUST both be valid based on the site certificate handed out by the server in the TLS exchange. Similarly, the domain name in the SRV query and the domain name in the target in the SRV record MUST both be valid based on the same site certificate. Otherwise, an attacker could modify the DNS records to contain replacement values in a different domain, and the client could not validate that this was the desired behavior or the result of an attack. I'm not sure what the phrase ...MUST both be valid based on the site certificate means. Does it mean that all possible domains must be present in the certificate? now imagine the follwing DNS lookups: sip:[EMAIL PROTECTED] ; order pref flags service regexp replacement IN NAPTR 50 50 s SIPS+D2T _sips._tcp.example.com. IN NAPTR 90 50 s SIP+D2T_sip._tcp.example.com IN NAPTR 100 50 s SIP+D2U_sip._udp.example.com. _sips._tcp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 01 5061 server1.example.com IN SRV 02 5061 server2.example.com _sip._tcp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 01 5060 server1.example.com IN SRV 02 5060 server2.example.com _sip._udp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 01 5060 server1.example.com IN SRV 02 5060 server2.example.com Finally, a TLS connection is made with server1.example.com. IF I understand RFC3263 correct, all of the above domains must be present in the certificate, but how to do this? 1. Should I put CN=example.com into the Subject and all other domains into the Subject Alternative Name? DNS=_sips._tcp.example.com. DNS=_sip._tcp.example.com DNS=_sip._udp.example.com. DNS=server1.example.com DNS=server2.example.com 2. Should I leave the Subject empty and put all domains into the Subject Alternative Name? 3. Why is it not sufficient to use only the domain example.com in the certificate (putting it into the subject field)? 4. Which SIP URIs should be used to check against the domains in the certificate (mutual proxy-proxy scenario)? Is it correct to check the domain in the request URI against the certificate of the receiving proxy, and check the domain in the From: URI against the certificate of the originating proxy? Thanks for any clarifications Klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] RE: SIPxchange
Try Asterisk.org klaus Abdul Lateef wrote: Hi Dale, I checked SIPxchange, But i found that it is commercial source. So Please let me know if there is any free Open Source for SIP IVR. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! for Good Watch the Hurricane Katrina Shelter From The Storm concert http://advision.webevents.yahoo.com/shelter ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] Re: [Serusers] Scalability in presence notification
Vincent Luba wrote: Hi, I need some advices about a design issue. I'm trying to figure out how to manage my presence information notifications, using RFC 3856. I have a situation in which end-to-end presence management wouldn't be suitable. Still I'm wondering if server based presence management is a good idea. My thought is that it present a scalability issue. The overall number of messages sent/received will be lower when using serverbased presence. Imagine a SIP client with modem connectin and 50 users on its buddylist: end2end: 50xNOTIFY to the SIP proxy + 50xNOTIFY to the SIP clients + 50 responses (200 ok) + 50 responses server based: 1xPUBLISH + 50xNOTIFY to the SIP clients +1 responses (200 ok) + 50 responses This will lower the traffic on the SIP proxy and on the UAC side. But of course, a presence server needs CPU power and has to store the presence locally in a database. Thus, as you see, this is a trade of. simple server and high network traffic vs complicated server and low traffic and centralized features (buddy list...) Additionally, you also have to consider SIP implementations. There are not thousands of SIP clients which supports all presence modes. Some clients supports end2end, some clients support server based. The same with SIP proxies. Conlusion. Get some clients and implement both scenarios and find the one which fits your needs. klaus The point is that having the presence server would have to handle every single SUBSCRIBE PUBLISH request then sends NOTIFY to every watchers. Is it an efficient way to handle presence in a network with hundreds online users, everyone having tens of contact on their list. Thank you for your comments Luba Vincent ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP win32 softphone
You can try http://www.enum.at/index.php?id=softphone klaus Abdul Lateef wrote: Hi all, Can anyone suggest me any open source for SIP dialer for win32 OS. Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP servers
sandeep kamath wrote: Hi, we are planning to evaluate SIP servers for Linux/BSD and Windows. We have tried the SER proxy for Linux and are quite happy with this. Are there other servers that you guys have used and would recommend for these platforms. I had a look at the ONDO server for windows but looks like it is a bit premitive. There is also openser, partysip and repro (resiprocate). regards klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Nokia Series 60 device
Gaurav Kheterpal wrote: Hi, You can refer to http://sw.nokia.com/id/ad2fe51c-afa0-42f1-b319-982bd0bd9f9a/Series_60_De veloper_Platform_2_0_Specification_v1_0_en.pdf for Series 60 compliance information. The Series 60 docs are available on http://www.forum.nokia.com/main/0,6566,010_40,00.html#documents Is there any particular aspect which you would like to know about the Series 60 compliance with RFC 3261 ? I have limited experience on using it but I found it *reasonably* compliant with SIP specs. Long time ago I did some testing. The only problem with the stack I can remember: I found out that in the Contact of a 200 OK response from a REGISTER, there must not be a q=...; parameter. If there is a q parameter, Nokia's SIP stack does not accept the 200 OK. Another interesting thing is, if support for IM+presence (MESSAGE, SUBSCRIBE, NOTIFY, PUBLISH) is planned. regards Klaus I hope this helps. my 2 cents, Gaurav -Original Message- ** Legal Disclaimer This email may contain confidential and privileged material for the sole use of the intended recipient. Any unauthorized review, use or distribution by others is strictly prohibited. If you have received the message in error, please advise the sender by reply email and delete the message. Thank you. ** From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: Thursday, August 11, 2005 1:46 PM To: Banibrata Dutta Cc: sip@ietf.org; sip-implementors@cs.columbia.edu Subject: Re: [Sip-implementors] Nokia Series 60 device Hi! Because it is usally much more easier to ask other developers for help. Do you now a contact at Nokia which we may ask about their SIP stack? Klaus Banibrata Dutta wrote: Why not ask Nokia ? -bd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rajesh Kadikar Sent: Thursday, August 04, 2005 2:16 AM To: sip@ietf.org; sip-implementors@cs.columbia.edu Subject: [Sip-implementors] Nokia Series 60 device Guys, I am not sure if this is the right forum for this question... But has anybody used the SIP stack that came out with Nokia Series 60? Do you know if it is RFC 3261 compliant? Thanks in advance Rajesh ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Nokia Series 60 device
Hi! Because it is usally much more easier to ask other developers for help. Do you now a contact at Nokia which we may ask about their SIP stack? Klaus Banibrata Dutta wrote: Why not ask Nokia ? -bd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rajesh Kadikar Sent: Thursday, August 04, 2005 2:16 AM To: sip@ietf.org; sip-implementors@cs.columbia.edu Subject: [Sip-implementors] Nokia Series 60 device Guys, I am not sure if this is the right forum for this question... But has anybody used the SIP stack that came out with Nokia Series 60? Do you know if it is RFC 3261 compliant? Thanks in advance Rajesh ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] Re: [Sip] Free B2BUA server.
Anil Bollineni wrote: Hi, Do you know any free B2BUA server that can I use for testing? Asterisk.org regards, klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Info abt Opensource
http://www.pernau.at/kd/voip/bookmarks-sip-rtp-ua.html regards, klaus sudhir kumar reddy wrote: Hi I would like to know about the SIP stack open source in C++ expect VOVIDA stack, which are avilable in net. thanks in advance Sudhir Kumar Reddy Thumma 09866060751 ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SMS using SIP
I have kannel connected to an SMSC via SMPP and using sipsak to generate MESSAGE messages. Works fine! SMSC--SMPP--kannel--htpp--apache+cgi+sipsak--SIP MESSAGE--SIP Client SMSC--SMPP--kannel--htpp--ser+perl script--SIP MESSAGE--SIP Client AFAIK 3GPP discusses the usage of SIP for SMS delivery. regards, klaus Kishoresowdi wrote: hi all, How can we support SMS using SIP , if at all we can? ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] can I add an E164 number as Reply-To address?
Hi all! Currently I'm playing around with SIP Gateways to make existing PBXs SIP/ENUM capable. I have the problem to signal valid CLI (Calling Line Identification) when interacting with traditional phone equipment. Please consider the following scenario: A B phone | | | PBX | | SIP/ENUM --- SIP phone | PSTN Party A has put a SIP Gateway between its PBX and the Telco to allow break-out, if the called party is also reachable via SIP. Party A uses SIP URI like sip:[EMAIL PROTECTED] in the From: header. Thus, If party B missed the call, it can call back to party A by calling sip:[EMAIL PROTECTED] But if B does not have a SIP phone, e.g. a handset as in the following setup, user B can not call back to user A. (The same problem is also in the first scenario, when B calls A and A wants to have a valid CLI on its display). A B phone phone | | | | | | PBX PBX | | | SIP/ENUM - | | | PSTN PSTN Usually, SIP gateways will extract the numbers from the from header. Thus, user B will see on the display of its normal phone: 66 The problem is, that B does not know the context in which the received number is valid. Some parties will use the extension as user part (sip:[EMAIL PROTECTED]) whereas others might use the national number (sip:[EMAIL PROTECTED]) or the E164 number (sip:[EMAIL PROTECTED]). Even worse, some will not use numbers at all - e.g. my SIP URI is sip:[EMAIL PROTECTED] Nevertheless, I have an E164 number which will be forwarded to my SIP Account. Thus, which means can be used to signal this E164 number to the callee? - The caller's SIP client can use a tel URI in the From header. This way I use my nice SIP URI when calling another SIP phone. Anyway I'm not aware of any SIP client which supports this. - There is the Remote-Party-Id header. This is an outdated draft, but still supported by most vendors. I can signal the phone number in Remote-Party-Id header, but still it does not allow me to set the context in which the number is valid (like a tel URI or like the type of number in ISDN). Are there other methods which I am not aware of to achieve dialing plan interoperability between SIP and SIP enabled traditional phones? What are the best practices for such scenarios? I like the idea of having my name in the From: header instead of a number. But when interacting with traditional phone equipment, signaling an additional valid number (CLI) which can be called back is essential. My first idea was having a Reply-To: header with a tel URI. best regards, Klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] PTime Question.
[EMAIL PROTECTED] wrote: Ramesh, My thoughts on your response / question. If the a= atribute is not grouped to a specific (valid) media codec then it is taken as applying to all. If the attribute is not valid for a certain media eg your example of ptime 40 not being valid for codec 4. Maybe a few things can happen in your scenario below. 1.codec 18 is matched | selected, ptime is applied and is acceptable - therefore session completes successfully using 18(ptime 40). 2.codec 4 is matched | selected, ptime is not acceptable - therefore UA returns a 488 Not Acceptable Here and session is not successful. 2a. codec 4 is matched | selected, ptime is not acceptable - UA discards 4 as it is not capable of the attributes associated with it and moves on to match | select the offered codecs in the m line (0,8,...) I don't like solution 2. Although the SDP is not correct, the client should use some intelligence to accept the session (...be liberal in what you accept). IMO the client should go on with the next supported codec. If codec 4 is the only supported codec, the client should use this codec, but with a valid ptime. regards, klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Equvalent of CANCEL for unanswered calls?
David Benoit wrote: Here's a bit of a strange questions, but one who's answer eludes me. signal. Any of the interoperability documents for PSTN gateways (over which I will likely have no control) only release with normal as a result of a CANCEL or BYE. I can't send a CANCEL because I'm the recipient of You can ask the gateway provider for a special cause code mapping. regards, klaus ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP Conference server
Rakesh Jain wrote: Hi, Could someone provide me pointers regarding 1. Any open source available for SIP conference server asterisk sems http://www.voip-info.org/tiki-index.php?page=sems regards klaus 2. Any open source (or algorithm) available for RTP Mixing Thanks in advance. Regards, Rakesh Yahoo! India Matrimony: Find your life partneronline. ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Establishing An Rtp Session
Hi Billy! I think SIP is the wrong protocol for sending MMS messages. AFAIK they are typically sent using SMTP or HTTP. regards, klaus Billy Carr wrote: Hey All! I am interested in using SIP and RTP stacks to send MMS messages. What is not clear is how to establish an RTP session once the SIP connection negogiations have successfully completed. Can I buy a vowel? :-) Thanks in Advance for your help. Bill Carr ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP - TEL URI Implementation Query
Hi! Gaurav.Kheterpal wrote: ... 2. Besides 'Request URI' and 'To:' header fields, can the TEL URI be present in any other INVITE header values (e.g. Via, Route Header, Contact etc). In other words, which fields in the INVITE request does the proxy scan for possible tel-sip translation using ENUM/ any other mechanism ? In case, there is no other field which contains this information, how is the TEL URI information propagated across intermediate proxies when the UA is configured with a outbound proxy having a defined route set. If the defined route set is proxy 1- proxy 2 in the below diagram, the request will be formed using: INVITE sip:proxy1 Route: sip:proxy2 To: tel:1234 this is a message in old style (RFC2???) with strict routers. According to 3261 (loose routers) the request URI contains always the final destination, e.g. the tel-uri. I think the message should look like this (not sure if the second route header is necessary): INVITE tel:1234 Route: sip:proxy1 Route: sip:proxy2 To: tel:1234 Intermediate proxies should forward the request according to the Route headers without any need to interpret the request URI. Only the final proxy has to interpret the request URI (tel uri). regards, klaus User A Proxy 1 Proxy 2 User B |||| | INVITE F1||| |---||| Does this mean that in such a case, the only parameter which contains the TEL URI is the 'To' field. As I understand, this field is never processed by the proxies unless they implement privacy/ RFC 3324(5) ?? I am current not aware of any proxy which is capable of handling TEL URI's, any information/ links regarding the same will be highly useful. Thanks in advance for your help. regards, Gaurav ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] multiple users on same ip
They can run on the same port - e.g. take a look at xlite (they are using the same port for all the configured users/proxies). regards, klaus [EMAIL PROTECTED] wrote: Hi, I'm interested in having multiple users registered from same machine e.g. [EMAIL PROTECTED] and [EMAIL PROTECTED] Do I have to setup the UAS to listen on different ports for each user e.g. [EMAIL PROTECTED]:5060 and [EMAIL PROTECTED]:5061 or is the messages sepparated in some other way? Thanks, //Jens ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP stack question
resiprocate/dum + jrtp http://pernau.at/kd/voip/bookmarks-sip-rtp-ua.html klaus m. smadi wrote: I need to write a mini sip client. Now, which RTP and SIP stacks do you recommend and why? I guess by using one of these stacks the need for implementing the SIP state logic is eliminated. thanks moe smadi ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Looking for a SIP softphone supporting TLS
Windows Messenger can be configured to use TLS (i've never tried it) klaus Shekar, Nagesh Soma (Shekar)** CTR ** wrote: All, I am looking for a softphone which supports TLS/SSL. Please provide pointers. Best Regards, Nagesh Shekar [EMAIL PROTECTED] 614-860-7664 ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] Symmetric RTP implementation
Hi Linda! I think this is what you are looking for: http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-comedia-08.txt regards, klaus Linda Xiao wrote: Hi all, Can anyone elaborate me on how to setup the symmetric RTP in the SDP part of a SIP Invite, or point me the web link of the relevant information. Thank you. Regards/Linda ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] which tools for building such apps?
Hi Cosmin! I recommend to not build your own SIP stack. Programming a stable SIP stack is not that easy - that is more work than a bachelors project. Use an existing SIP stack and build your application on top of it. There are several OO SIP stacks like resiprocate, sipX... which offers all the low level SIP functionality. Furthermore I would not waste time and develop a SIP server/proxy as they are also available (e.g. ser). There is still lot of work left by combining voice/IM/presence under one GUI. At http://www.pernau.at/kd/voip/bookmarks-sip-rtp-ua.html you can find links to several stacks and SIP applications/server which may help you. regards, Klaus Cosmin Banu wrote: I really haven't thought about developing something novel, although I think I should. Since I will put alot of effort into making the project (I am an undergraduate student so I suppose this corresponds to a bachelors project), I was thinking of reusing the knowledge gained for my future career, and SIP looks promising. As for the point of the project, I think it's about testing my design and implementation capabilities (I will use C++ as development language, not C, so I can fiddle around with object oriented design). Regards, Cosmin On Tue, 06 Jul 2004 11:18:48 -0400, Paul Kyzivat [EMAIL PROTECTED] wrote: I don't believe you mentioned what scale of project this is for you - bachelors project, masters, or PhD. Building a complete SIP stack that is anything but a toy is a very large undertaking. (Don't be deceived by the claims that it is a simple text based protocol.) It is possible to do something simple that handles a subset of sip, and that works with itself in simple cases. or perhaps even works with one or two selected devices. Implementing SIP today is not novel, it has been done many times. You should probably consider what the point of the project is - to show something novel, to show how much code you can write in a limited amount of time, to demonstrate end user functionality, or what? Paul ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP Conference Implementation ?
Try asterisk (the meetme application) or the conference plugin of sems (iptel.org) klaus Shahed Moolji wrote: Hello, I have heard on the nist-sip mailing list, that someone on this list had implemented a SIP conference server, including the mixing of RTP streams. Does anyone know where I can find any information related to this ? I am more interested in an opensource implementation of an RTP audio stream mixer, as I have heard from others that JavaSound / JMF solutions are not scalable, and I have been unable to locate any other implementation. Thanks Shahed. ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] Re: [Serusers] Forking Proxy
UA1 behaves wrong - per RFC 3261 the dialog will be established by the 200 OK message (not by 180), therefore the to-tag from the 200 OK must be used. It makes no sense to deal with such wrong implementations in the SIP proxy. Which client is UA1? Klaus Jason Penton wrote: Hi All I have a question about the CORRECT operation of UA when making a call via a forking proxy: Lets say UA1 calls [EMAIL PROTECTED] and user [EMAIL PROTECTED] is available at UA2 and UA3 i.e. the Proxy forks the request. UA1 Forking UA2 UA3 Proxxy 1 |--INVITE--|| | 2 | |INVITE--| | 3 | |---INVITE-| 4 | |---RINGING(totag=1234)--| | 5 |---RINGING(totag=1234)|| | 6 | |--RINGING(totag=5768)-| 7 |---RINGING(totag=5678)|| | | | | | | | | | NOW UA3 WILL ANSWER | |200 OK (totag=5678)---| 8 |-200 OK (totag=5678)--|| | 9 |---ACK (totag=1234)---|| | 10 | |--ACK (totag=1234)| AT THIS STAGE UA 3 IGNORES THE ACK AS IT DOES NOT CORRESPOND TO ORIGINATING 200 OK AND THE CALL IS NOT SETUP * UA1 is using the to-tag of the first 180 RINGING it received (frame 5) no matter what the to-tag in the 200 OK is * My question here is: who is in the wrong The proxy or UA1? * Should the proxy change the to-tag of the ACK before forewarding it to UA3??? Any help/guidance would be much appreciated Jason Penton Rhodes University Grahamstown South Africa ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] some help needed
There are other applications which are (IMHO) much more easy to configure and to use than vocal. So, if you don't need a certain feature of vocal I would recommend to start with the following applications: sip proxy: ser (iptel.org) SIP client: windows: xlite (xten.com) linux: kphone wirlab.net(kphone) [EMAIL PROTECTED] wrote: Hi, I am cuurently pursuing my masters inElectrical Engineering from the University of South Florida. I am deeply interested in this topic, being a novice I hope somebody could help me with my query. I have a few doubts regarding the SIP software given in site www.vovida.org,and would be glad to recieve some guidance regarding these issues: I get the main window GUI for the SIPset software. It says that after entering a few basic configuration parameters, you get registered to make calls. Here since we are trying to make peer-peer call, without the proxy server, I followed the instructions accordingly. However the main window does not show the registered icon. For machine one, with user name kept say 1000 and IP address 100.100.100.101, the log messages says error 404 and the user not found. For machine two, with user name say 1001 and IP address 100.100.100.100, the log message gives messages like rtpmap, audio, application SDP process Does this mean that for the second machine it is working, though not registered? Thank you Sharda Divekar ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP parser library under Linux
You can use open source stacks, but they are bad documented. For free stacks take a look at: http://www.ict.tuwien.ac.at/darilion/bookmarks-sip-rtp-ua.html section: SIP Stacks regards, Klaus Maxim wrote: Hello! I am Maxim from Russia. Work for Raskat telecommunication company. We're looking for a software parser of SIP protocol. We would like it to be C or C++ source code under Linux with documentation to use it. Is there available software like this ? Thank you in advance, Maxim ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP programming in C++.
linphone klaus Tae-Sam Kim wrote: I installed oSIP in my linux box. Do you any places I can find tutorial SIP implemenation (application using SIP/TCP and SIP/UDP) in C or C++? I am very new on SIP. Also I am looking for some exmaples codes like client-server implementation using SIP. Thanks. Tae-Sam From: kaiduan xie To: Tae-Sam Kim Subject: Re: [Sip-implementors] SIP programming in C++. Date: Wed, 18 Feb 2004 13:40:28 -0500 (EST) Tae Sam, I have some working knowledge with VOVIDA, oSIP, SER and KPhone. The following is my understanding: VOVIDA (C++): A Very very huge library and sip stack, also many full-blown server, like UA, Marshal server, Redirect Server, Provision Server, B2BUA, SNMP server. It doesnot comply with RFC 3261. Resiprocate (C++) http://resiprocate.sourceforge.net A new sip stack designed by the same guys in VOVIDA, it fully comply with RFC 3261, but it is just a C++ SIP stack. SER (C only): A very fast and small SIP proxy server, using many special skills. It's SIP stack cannot be used in other place and other application. KPhone's SIP stack (C++): Purly based on QT library, no multi-thread support. oSIP (C): A great sip stack, very efficient, small footprint, and very very readable. You can read the sourcecode and the collate it with the standard! Hope that helps, kaiduan --- Tae-Sam Kim wrote: - Vovida's URL doesn't work, and NIST's SIP has only Java. Do you know any other place for SIP programming in C++ or C ? I've searched lots places, but I couldn't find what I wanted. Thank you. Tae-Sam From: a a To: Tae-Sam Kim , [EMAIL PROTECTED] Subject: Re: [Sip-implementors] SIP programming in C++. Date: Tue, 17 Feb 2004 16:25:38 -0800 (PST) Hi, There are many freely available, source code public implementations of SIP. Check out the following links: Vovida's SIP : http://www.vovida.org/ NIST's SIP: http://snad.ncsl.nist.gov/proj/iptel/ Should you do some search on the net, you could fine more. Cheers... --- Tae-Sam Kim wrote: - I am looking for a tutorial that explain how client-server applications(proxy) can be written in C++ using SIP. Also, I want to download all C/C++ libraries for implemenation using SIP. Where can I download free version of the libraries and installation instruction? Thanks. Tae-Sam - Get fast, reliable access with MSN 9 Dial-up. Click here for Special Offer! ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html - Stay informed on Election 2004 and the race to Super Tuesday. ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors __ Post your free ad now! http://personals.yahoo.ca Take off on a romantic weekend or a family adventure to these great U.S. locations. http://g.msn.com/8HMAENUS/2746??PS= ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP programming in C++.
Hi! Piotr Gutkowski wrote: Hello! I'm starting my work with SIP too. I have downloaded few implementations and I'm considering which one will be good. KPhone is Suse dedicated written in C++ but looks not bad for newbie because of small size. Maybe someone used it with Red Hat 9.0? No problems? kphone is not Suse dedicated. kphone works an all linux distribution. It requires the qt library. Vovida is huge but maybe has got the power (incl. MGCP, RTP, etc.). IMHO vovida is a bad starting point - much to big. NistSIP Ubiquity are Java oriented. BTW there is small SIP tutorial on Nist site. I'm getting information about SIP from Henning's site and from Sipcenter. My friend suggested me to use OPAL (part of OpenH323 http://www.openh323.org/) that's H323 != SIP and I'm thinking about Asterisk working as a server - http://www.asterisk.org/) If You have any experience with these implementations I will be glad to hear your opinion. I you wan't to setup a SIP based infrastructure, I suggest iptel's SIP proxy (open source) (runs on unix-platforms). As client I suggest xlite for windows and kphone for linux. If you wan't to develop your own SIP application, it depends on the operating system. Linux: kphone can be easily extended, or you just use the SIP stack (dissipate) from kphone and write your own application on top of it. Windows: Use osip/exosip or resiprocate as SIP stack and write your own application. At the moment I'm not aware of an open source SIP client for Windows. Nevertheless theres a SIMPLE client included in resiprocate. For RTP you can also choose from a lot of available open source RTP stacks. A overview of available stacks (C/C++) can be find at: http://www.ict.tuwien.ac.at/darilion/bookmarks-sip-rtp-ua.html regards klaus Greetings Piotr Tae-Sam Kim wrote: I am looking for a tutorial that explain how client-server applications(proxy) can be written in C++ using SIP. Also, I want to download all C/C++ libraries for implemenation using SIP. Where can I download free version of the libraries and installation instruction? Thanks. Tae-Sam ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP video phone call flow
It's the same as for a normal phone call. The only difference is the SDP in the INVITE message. It also contains the description for the video stream (IP address and port, codecs...) Klaus Jayakara N wrote: Hi all, Can anybody give me callflow for video phone? I need to create two RTP connections, one is for audio and other is for video. Can anybody tell me how SIP message for initiates these two connection? Can it initiated by one INVITE message? or I need to have two saparate INVITE message one is for audio and other is video? Thanks in advance. Regards Jay -- Domain Hosted and Supported by Hindunet Inc. (http://www.hindunet.com/) -- Domain Hosted and Supported by Hindunet Inc. (http://www.hindunet.com/) ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] SIP programming in C++.
c: osip c++: dissipate (the stack of kphone) or resiprocate all of them lack documentation (I think osip has one, but its not up2date) klaus Tae-Sam Kim wrote: I am looking for a tutorial that explain how client-server applications(proxy) can be written in C++ using SIP. Also, I want to download all C/C++ libraries for implemenation using SIP. Where can I download free version of the libraries and installation instruction? Thanks. Tae-Sam Get fast, reliable access with MSN 9 Dial-up. Click here for Special Offer! http://g.msn.com/8HMAENUS/2752??PS= ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] RE: Welcome to the Sip-implementors mailing list
For NAT traversal, there is no need to change the callid! Klaus Anil Bollineni wrote: Hi, I have a question regarding callid. Two UA's behind firewall generate call id's in their INVITE messages as below. UA1: [EMAIL PROTECTED] UA2: [EMAIL PROTECTED] After firewall do the NAT translation in the callid's UA1: [EMAIL PROTECTED], UA2:[EMAIL PROTECTED] Whether this is permissible according SIP specification. Thanks, Anil This email contains material that is confidential. The content of this email is for the sole use of the intended recipient(s). Any review or distribution by persons other than the intended recipient(s) without the express permission of NetScreen Technologies, Inc. is strictly prohibited. If you are not the intended recipient, please contact the sender and delete/destroy all copies of this email and any related attachments. NetScreen does not guarantee the accuracy or completeness of third party materials or information. ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] how to invite someone to a conference?
Hello! What is the best practice to invite participants to a conference which is mixed on a central conference server (a conference at the conference server will be indicated by the username, eg. sip:[EMAIL PROTECTED])? For example: A has an active call with B, then A wants to invite C to set up a conference, which will be served by the conference server S? How does A signal to B that it should hang up and reconnect to S (sip:[EMAIL PROTECTED])? How does A signal to C that it should call S (sip:[EMAIL PROTECTED])? Or is it better to stop the call with B and tell S to invite B and C? thanx for your proposal Klaus ___ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
