Hi Nithyakumar, Please find reply to your queries raised
Record route: Record route is not mandatory header in all SIP request Record route header is add by proxies Record route header field is inserted by proxies in a request to force future requests in the dialog to be routed through the proxy Specified in RFC 3261 VIA: VIA header is mandatory header in all SIP request VIA header add by originator VIA header field indicates the transport used for transaction and identifies location where the response is to be sent Specified in RFC 3261 Regards manmohan singh bisht On Wed, 24 Jan 2007 [EMAIL PROTECTED] wrote : >Send Sip-implementors mailing list submissions to > [email protected] > >To subscribe or unsubscribe via the World Wide Web, visit > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors >or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > >You can reach the person managing the list at > [EMAIL PROTECTED] > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Sip-implementors digest..." > > >Today's Topics: > > 1. Re: Difference bt Via nd Record-Route (Nithyakumar.R) > 2. Re: Difference bt Via nd Record-Route (Gary Cote) > 3. CODEC usage (Andrew Chalk) > 4. Adds UUI in SIP signalling (JaeSeung Song) > 5. Re-Invite (Sambit Pooja) > 6. Re: Re-Invite (Gary Cote) > 7. Call status when using PBX as gateway (Bruce Atherton) > 8. Re: Call status when using PBX as gateway > ([EMAIL PROTECTED]) > 9. Re: Call status when using PBX as gateway (Bruce Atherton) > 10. Re: Call status when using PBX as gateway > ([EMAIL PROTECTED]) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Mon, 22 Jan 2007 11:14:42 +0530 > From: "Nithyakumar.R" <[EMAIL PROTECTED]> >Subject: Re: [Sip-implementors] Difference bt Via nd Record-Route >To: "'Gary Cote'" <[EMAIL PROTECTED]>, "'Justin Lu'" > <[EMAIL PROTECTED]> >Cc: [email protected] >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=us-ascii > >Hi Gary, > > "The Record-Route header governs the path that subsequent requests >within the dialog will take.." > > When Record Route is enabled in proxies it will update Via header..if so >then Please give more information for above mention statement..Thanks in >advance.. > >With regards, >Nithyakumar.R > > >-----Original Message----- > From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Gary Cote >Sent: Wednesday, January 17, 2007 7:21 PM >To: Justin Lu >Cc: [email protected] >Subject: Re: [Sip-implementors] Difference bt Via nd Record-Route > >The Via header governs the path that the response will take. > >The Record-Route header governs the path that subsequent requests >within the dialog will take. > >So the UAS doesn't use the Record-Route header to determine where to >send a response. > >-- >Gary Cote >www.awardsolutions.com >_______________________________________________ >Sip-implementors mailing list >[email protected] >https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > >------------------------------ > >Message: 2 >Date: Mon, 22 Jan 2007 08:40:47 -0600 > From: "Gary Cote" <[EMAIL PROTECTED]> >Subject: Re: [Sip-implementors] Difference bt Via nd Record-Route >To: "Nithyakumar.R" <[EMAIL PROTECTED]> >Cc: [email protected] >Message-ID: > <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >On 1/21/07, Nithyakumar.R <[EMAIL PROTECTED]> wrote: > > > > When Record Route is enabled in proxies it will update Via header..if so > > then Please give more information for above mention statement..Thanks in > > advance.. > > > >That's not entirely correct. The proxy will update the Via header >regardless of whether Record-Route is enabled. > >Read rfc3261, section 16.6 on proxy request forwarding and you will >see that the proxy may optionally add a Record-Route header (as you >say, if it's enabled), but must always add a Via header. > >In other words, the two headers operate independently of each other. > >-- >Gary Cote >www.awardsolutions.com > > >------------------------------ > >Message: 3 >Date: Mon, 22 Jan 2007 14:00:55 -0600 > From: "Andrew Chalk" <[EMAIL PROTECTED]> >Subject: [Sip-implementors] CODEC usage >To: <[email protected]> >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset="us-ascii" > >I wondered: Does anybody know how much VoIP (or SIP) traffic is carried by >each of the major media CODECs? Presumably someone keeps tabs on this. > > > >A distinction between on-LAN and off-LAN traffic would also be useful. > > > >Many thanks. > > > > > > > >------------------------------ > >Message: 4 >Date: Tue, 23 Jan 2007 19:17:51 +0900 > From: "JaeSeung Song" <[EMAIL PROTECTED]> >Subject: [Sip-implementors] Adds UUI in SIP signalling >To: <[email protected]> >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset="ks_c_5601-1987" > >Hi, > >Where can I find information on signalling the UUI Information Element in SIP? >I want to implement the following scenario; > 1. The SIP AS adds UUI into the SIP INVITE message and sends it to the > MGCF. > 2. The MGCF performs interworking between the IMS and the CS. > The MGCF just graps the UUS1 from the SIP INVITE message and maps into > proper ISUP UUI parameter. > 3. Resulting ISUP message contains UUI. > >Some of RFC and draft menitoned the way forward. >For example, >draft-johnston-sipping-cc-uui-01 describes and recommands to use User-to-User >header. >But it's still draft. As the recommandation describes, a new SIP header field >will be defined and >move to a SIP working group document? > >Or Is there any standard way? > >BR, >Anthony Song > >------------------------------ > >Message: 5 >Date: Tue, 23 Jan 2007 07:58:04 -0800 (PST) > From: Sambit Pooja <[EMAIL PROTECTED]> >Subject: [Sip-implementors] Re-Invite >To: [email protected] >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=iso-8859-1 > >Hi all > > Why it is not preferable to send Re-INVITE in a pre established session ? > > Thanx > > Sam > > >--------------------------------- >Don't be flakey. Get Yahoo! Mail for Mobile and >always stay connected to friends. > >------------------------------ > >Message: 6 >Date: Tue, 23 Jan 2007 11:29:47 -0600 > From: "Gary Cote" <[EMAIL PROTECTED]> >Subject: Re: [Sip-implementors] Re-Invite >To: "Sambit Pooja" <[EMAIL PROTECTED]> >Cc: [email protected] >Message-ID: > <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Sending a re-INVITE within an established dialog is a perfectly valid >operation. >Please clarify your question or explain the context. > >I doubt we can help you, otherwise. > >-- >Gary Cote >www.awardsolutions.com > > >------------------------------ > >Message: 7 >Date: Tue, 23 Jan 2007 11:33:28 -0800 > From: Bruce Atherton <[EMAIL PROTECTED]> >Subject: [Sip-implementors] Call status when using PBX as gateway >To: [email protected] >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=us-ascii; format=flowed > >I am trying to write a program that makes a SIP connection to a PBX and >asks the PBX to dial out on a PSTN line. Once someone answers, the >program plays a message to the person phoned to let them know about a >problem they need to fix. I'd like to make the program work with as many >PBXs as possible. > >My problem is that I don't want the PBX to respond "200 OK" to the >INVITE until the PSTN line has actually been picked up. In other words, >the SIP call status should be based on the PSTN call status. I'm not >worried about "180 Ringing" state as it is difficult to detect ringing >on the PSTN, but "200 OK" must wait for the voltage drop that indicates >a connection on the PSTN. > >The reason I need this is that otherwise, the program starts playing the >message about what needs fixing while the PSTN phone is still ringing. >Also, the program can never tell whether a call was received by someone >or not. So it doesn't know whether to go further with the call tree. > >Is there a standard way in SIP to accomplish this? Is there anything my >program can do to instruct a SIP-enabled PBX to make the SIP state >mirror the PSTN state the call is gatewayed to? Or is there something it >can watch for in its communications with the PBX that is independent of >the SIP call status, but will let it know when the PSTN line goes to a >connected state? > >If there is no standard way to accomplish this, then does anyone know >which PBXs can be configured to act this way? A generic solution would >be best, but if need be I'll deal with the most popular PBXs >individually. My guess is that PBXs which advertise that they can act as >both a SIP Proxy and a SIP Gateway are the ones that have the potential >to deal with call state in this way, but I'm finding it hard to find >information on the subject. > >I am really struggling to get a handle on how to deliver this >functionality. Any help is greatly appreciated, even if it is just >search terms I should be targeting for researching this. > > >------------------------------ > >Message: 8 >Date: Tue, 23 Jan 2007 16:07:31 -0500 > From: [EMAIL PROTECTED] >Subject: Re: [Sip-implementors] Call status when using PBX as gateway >To: [email protected] >Message-ID: <[EMAIL PROTECTED]> > > From: Bruce Atherton <[EMAIL PROTECTED]> > > My problem is that I don't want the PBX to respond "200 OK" to the > INVITE until the PSTN line has actually been picked up. In other words, > the SIP call status should be based on the PSTN call status. I'm not > worried about "180 Ringing" state as it is difficult to detect ringing > on the PSTN, but "200 OK" must wait for the voltage drop that indicates > a connection on the PSTN. > >Remember that the endpoint user agents in the SIP dialog are your user >agent and the PSTN gateway device -- the PBX just passes SIP messages >between them. > >Generally speaking PSTN gateways only return 200 to an INVITE when the >PSTN reports that the other end of the PSTN call has answered. > >Dale > > >------------------------------ > >Message: 9 >Date: Tue, 23 Jan 2007 13:41:31 -0800 > From: Bruce Atherton <[EMAIL PROTECTED]> >Subject: Re: [Sip-implementors] Call status when using PBX as gateway >To: [email protected] >Message-ID: <[EMAIL PROTECTED]> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >That is really good news. So the problem must just be with Asterisk, >which happens to be the PBX I am testing with. I will take the issue up >with the developers there. > >Thanks a lot for your help. I was worried that I was looking at a much >larger problem. > >Dale.Worley-at-comcast.net |SIP Implementors| wrote: > > From: Bruce Atherton <[EMAIL PROTECTED]> > > > > My problem is that I don't want the PBX to respond "200 OK" to the > > INVITE until the PSTN line has actually been picked up. In other words, > > the SIP call status should be based on the PSTN call status. I'm not > > worried about "180 Ringing" state as it is difficult to detect ringing > > on the PSTN, but "200 OK" must wait for the voltage drop that indicates > > a connection on the PSTN. > > > > Remember that the endpoint user agents in the SIP dialog are your user > > agent and the PSTN gateway device -- the PBX just passes SIP messages > > between them. > > > > Generally speaking PSTN gateways only return 200 to an INVITE when the > > PSTN reports that the other end of the PSTN call has answered. > > > > Dale > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > >------------------------------ > >Message: 10 >Date: Tue, 23 Jan 2007 21:52:58 -0500 > From: [EMAIL PROTECTED] >Subject: Re: [Sip-implementors] Call status when using PBX as gateway >To: [email protected] >Message-ID: <[EMAIL PROTECTED]> > > From: Bruce Atherton <[EMAIL PROTECTED]> > > That is really good news. So the problem must just be with Asterisk, > which happens to be the PBX I am testing with. I will take the issue up > with the developers there. > >Well, I know that the open-source sipX PBX doesn't have this problem >-- when you use it with several popular brands of PSTN gateway. But >sipX is "pure SIP", the PBX really is a proxy, and the two endpoints >are swapping status info, so if the gateway is well-behaved, your UA >will see the right things. > >Asterisk isn't done that way, because SIP was retrofitted onto another >protocol (IXA), so your UA talks SIP to Asterisk, which talks SIP to >the gateway; really two SIP dialogs which together do the job. That >being said, though, I would be surprised if Asterisk sends your UA a >200 before the gateway sends Asterisk a 200, to avoid exactly the >problems you mention. > >I'd check with the Asterisk developers, and make sure you tell them >which the gateway device you're using. You're running into something >that has to be unusual. > >Dale > > >------------------------------ > >_______________________________________________ >Sip-implementors mailing list >[email protected] >https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > >End of Sip-implementors Digest, Vol 46, Issue 19 >************************************************ _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
