Subject: [Sip] How to exchange mid-call information between two UAs?

From: 邹军 <[email-masked]>
Date: Mon, 8 Aug 2005 15:54:58 +0800 (CST)

Hi!
There is a problem, I need a help.
 
In a session, if a UA found there is a little congestion in the IP network, and it want to modify the initial parameters of session to low the rate of RTP by changing codec as to keep call on, then there may be a negotiation between two UAs. 
 
The problem is how exchange such mid-call infomation between two UAs in a session? 
INFO is designed to carry mid-call information along the SIP signaling path during the session, but not used to change the state of SIP calls, or the parameters of the sessions SIP initiates.
 
"In addition, the SIP re-INVITE can be used during a session to change
the characteristics of the session. This is generally to change the
properties of media flows related to the session or to update the SIP
session timer."
 
What's good method?
 
Thanks for any advice and suggestion!

---------------------------------
DO YOU YAHOO!?
  雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use [email-masked] for questions on current sip
Use [email-masked] for new developments on the application of sip

---

From: "James M. Polk" <[email-masked]>
Date: Mon, 08 Aug 2005 10:51:20 -0500

Use reINVITE for any changes to the parameters of a dialog - which you are 
describing.

INFO is for additional information about the session. Look at the example 
usages in RFC2976 as guidance here (signal strength, etc).

At 03:54 PM 8/8/2005 +0800, =?gb2312?q?=D7=DE=BE=FC?= wrote:
>In a session, if a UA found there is a little congestion in the IP 
>network, and it want to modify the initial parameters of session to low 
>the rate of RTP by changing codec as to keep call on, then there may be a 
>negotiation between two UAs.
>
>INFO is designed to carry mid-call information along the SIP signaling 
>path during the session, but not used to change the state of SIP calls, or 
>the parameters of the sessions SIP initiates.
>
><http://cn.mail.yahoo.com/mail_alert/promo1.html>ÑÅ»¢ÓÊÏä³¬Ç¿ÔöÖµ·þÎñ£­2G³¬´ó¿Õ¼ä¡¢pop3ÊÕÐÅ¡¢ÎÞÏÞÁ¿ÓÊ¼þÌáÐÑ 
>

cheers,
James

                                 *******************
                 Truth is not to be argued... it is to be presented.