Dipti,
No matter what the caller (sip phone user) does, the sip phone is expected to remain conformant to 3261. How it does this is its problem, but sending a reinvite while the prior invite is still incomplete is not an acceptable behavior.
Using UPDATE is an option for it. But of course that might not be implemented at one end or the other, so it may not be a sufficient solution.
Another solution is for the phone to simply wait until the invite completes. In the meantime it can suppress the transmission of media to/from the user as appropriate.
Paul
dipti wrote:
hi.
I completely agree with you, but what if the caller (a SIP phone user) while waiting for the call to be connected press the mute button or hold button. This will lead to a reinvite being send. How will the proxy handle it.
Thanks Dipti
-----Original Message----- From: Paul Kyzivat [mailto:[EMAIL PROTECTED] Sent: Friday, December 03, 2004 1:08 AM To: dipti Cc: [EMAIL PROTECTED] Subject: Re: [Sip-implementors] Mute
Dipti,
It is not permissible to send a reinvite before the prior invite has completed. So your scenario is invalid.
To fix, either you must wait for the first invite to complete, or if you must make the change earlier then you can potentially use an UPDATE.
Paul
dipti wrote:
Hi,
My scenario is
UA ---------------> Proxy |---> Invite --------180 ringing <---- UA -------> mute (reinvite) |--->
----200 OK <-----------
In this scenario, whats the behavior of the Proxy, when UA sends a mute while proxy is trying to connect a call.
Thanks and Best regards, Dipti Jain
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