Hi In SIP RFC 3261 its mentioned about the time , and it tells how long a proxy should wait after a Provisional 1xx response and more over it also tells that time can be implementations specific. regds Sumit Jayaswal.
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Re: Status checking between IP Phones and a proxyserver (Poojan Tanna) 4. sip + umts network (anne-marie boustany) ---------------------------------------------------------------------- Message: 1 Date: Fri, 28 Jan 2005 10:18:50 -0500 From: "Mariano-Julio Stokle" Subject: RE: [Sip-implementors] Timer after ringing To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain Yes, it is implementation specific. F.i. you could let users to determine it or just the admin. Mariano Stokle Marketing Manager - Enterprise Networks Larrea 1079 (C1117ABE) Buenos Aires Argentina [EMAIL PROTECTED] T +54 (11) 4827-7237 F +54 (11) 4827-7203 mailbox 7237# -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, January 27, 2005 9:07 PM To: [email protected] Subject: [Sip-implementors] Timer after ringing Hi, all How much time should the proxy server have after getting 180 RINGING method from an IP phone ? >From the "draft-ietf-sipping-service-examples-07, 2.9 Call Forwarding - No Answer", it's saying " /* B1 ring until a configuration timer expires in the Proxy. The Proxy sends Cancel and proceeds down the list of routes. */". I could not find what would be the "configuration timer". Is that just implementation specific ? Thanks, Jun _______________________________________________ Sip-implementors mailing list [email protected] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ------------------------------ Message: 2 Date: Fri, 28 Jan 2005 11:43:03 -0800 From: Poojan Tanna Subject: Re: [Sip-implementors] Timer after ringing To: Hyoungjoon Park Cc: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii Hi, Ringing time basically equals to Timer B with a value of 64*T1 seconds till the transaction times out. The default value of T1 is 500 msec so IP Phone can ring for 32 sec. There are two way to control the ringing time: 1. Either you increase/decrease the T1 value provided that the value is configurable. 2. Have a new propriety 'ringing timer' on the box and keep it configurable. After receiving 180/183 response, the new 'ringing timer' will start and will allow IP Phone to ring till the configured value. The second method is more desirable because changing the T1 value has other repercussions as Timer A also uses T1 value. Thanks, Poojan. Hyoungjoon Park wrote: > > Hi, all > How much time should the proxy server have after getting 180 RINGING > method from an IP phone ? > > From the "draft-ietf-sipping-service-examples-07, 2.9 Call Forwarding - > No Answer", it's saying " /* B1 ring until a configuration timer expires in > the Proxy. The Proxy sends Cancel and proceeds down the list of routes. */". > > I could not find what would be the "configuration timer". Is that just > implementation specific ? > > Thanks, > Jun > _______________________________________________ > Sip-implementors mailing list > [email protected] > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors -- ------------------------------------------------------------------------- Poojan Tanna E-Mail: [EMAIL PROTECTED] Phone Office : 510-747-5282 Home : 510-569-8820 ------------------------------------------------------------------------- ------------------------------ Message: 3 Date: Fri, 28 Jan 2005 11:52:13 -0800 From: Poojan Tanna Subject: Re: [Sip-implementors] Status checking between IP Phones and a proxyserver To: Hyoungjoon Park Cc: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii Hi, I think SIP Session Timer answers your question. The SIP Session Timer draft is available at the following location: http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-15.txt Session Timer will include one entity playing the role of Refresher and another Refreshee. The Refresher will keep sending the refresh re-Invite request to the Refreshee depending upon the configured/negotiated time once the call is UP. Thus if Refreshee does not receive the re-Invite till the Session Timer expires then it bring down the call. Thanks, Poojan. Hyoungjoon Park wrote: > > Hi all, > > I have some questions about checking status of IP phone. > > I wonder if there's any way to keep tracking of the IP phone's status. > In case when one of phones might be power-off or transport links beetween > phones and the proxy server got disconnected, the server could not figure > out the status of phones. > Knowing of the status of phone - especially dialog status - would be > very important in terms of "call logging or accunting". > > I've considered using Subscribe/Notify methods but I'm not sure what > would be the practical solution for this. > Do I have to use some kinds of proprietary messages such as a Skinny > from Cisco ? > > Thanks > Jun > > _______________________________________________ > Sip-implementors mailing list > [email protected] > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors -- ------------------------------------------------------------------------- Poojan Tanna E-Mail: [EMAIL PROTECTED] Phone Office : 510-747-5282 Home : 510-569-8820 ------------------------------------------------------------------------- ------------------------------ Message: 4 Date: Fri, 28 Jan 2005 21:34:28 +0000 From: "anne-marie boustany" Subject: [Sip-implementors] sip + umts network To: [email protected] Message-ID: Content-Type: text/plain; charset="us-ascii" An HTML attachment was scrubbed... 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