Well, the full text of that section refers to media failure, so I would assume 
that it is not so relevant to situations involving just "a little congestion":

 The behavior of a UA on detection of media failure is a matter of
   local policy.  However, automated generation of re-INVITE or BYE is
   NOT RECOMMENDED to avoid flooding the network with traffic when there
   is congestion.


-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED]
Sent: 09 August 2005 07:28
To: [email protected]
Subject: [Sip-implementors] How to exchange mid-call infomation between twoUAs 
in a session?



 

Quote from "14 Modifying an Existing Session" in RFC3261:

"However, automated generation of re-INVITE or BYE is NOT RECOMMENDED to avoid 
flooding the network with traffic when there  is congestion."

 

Is it OK to change the parameters of a dialog using such automated generation 
of re-INVITE?





>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 to 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!




                
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