Hello Sidhartha ,
You should be ready to receive media on any of the codecs you offered and also 
be prepared even if the other side dynamically switches between the codecs 
without additional signaling :)
Regards ,
Sayan

________________________________

From: Siddhardha Garige [mailto:[EMAIL PROTECTED]
Sent: Friday, May 26, 2006 11:17 AM
To: Nataraju Basavaraju; Sayan Chowdhury (WT01 - IP-Multimedia Carrier & Ent 
Networks); [email protected]
Subject: RE: [Sip-implementors] Query on when to open RTP ports 
duringoffer/answer


Hi Nataraju,

This is the sample invite send out by my UA.

INVITE sip:[EMAIL PROTECTED];user=phone 
<http://mail.yahoo.com/config/login?/sip:[EMAIL PROTECTED];user=phone>  SIP/2.0
Via: SIP/2.0/UDP 192.168.252.98:5061;branch=z9hG4bK31363c37f
Call-ID: [EMAIL PROTECTED] <http://us.f513.mail.yahoo.com/ym/[EMAIL PROTECTED]>
From: Anonymous;tag=69fd5c35
To:
CSeq: 1 INVITE
Contact:
Supported: 100rel
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIF
Y,MESSAGE,REFER
Content-Length: 311
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 1481 1481 IN IP4 192.168.252.98
s=Sip Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 11944 RTP/AVP 8 0 4 18 97 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:96 red/8000
a=fmtp:97 0-15
a=fmtp:96 97/97

I can open a socket and start listening on 11944 for RTP data. But how do i 
knwo what codec callee is doing with out getting a response for him?

-Sid


Nataraju Basavaraju <[EMAIL PROTECTED]> wrote:

        Hello All,

        Assume the case where in callee wants to play some music to caller, and 
the provisional responses were lost in the network.

        In this scenario, the music will not be played to caller, if the port 
was not opened when the INVITE with SDP was sent out... Hence I suggest open 
the RTP port for listening soon after the INVITE is sent out...

        Regards,
        Nataraju A.B.

        -----Original Message-----
        From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED]
        Sent: Thu 5/25/2006 8:33 PM
        To: [EMAIL PROTECTED]; [email protected]
        Subject: Re: [Sip-implementors] Query on when to open RTP ports 
duringoffer/answer


        Hello Siddhardha/All ,
        The offer answer RFC says that you should be ready to receive media as 
soon as you send the offer.
        That is where my doubt comes in ...
        If I start receiving my media before any answer from the other side , 
how do I know for sure that it's actually the intended called party who is 
sending me the packets.
        Also ,after I receive the answer if the called party plays the media 
from a different IP address than the one he is using to receive media , how do 
I know that it's actually the called party who is playing me the media , I 
looked into SRTP , but that does not seem to have a solution for this ...
        Regards ,
        Sayan


        ________________________________

        From: Siddhardha Garige [mailto:[EMAIL PROTECTED]
        Sent: Thursday, May 25, 2006 8:17 PM
        To: Sayan Chowdhury (WT01 - IP-Multimedia Carrier & Ent Networks); 
[email protected]
        Subject: Re: [Sip-implementors] Query on when to open RTP ports during 
offer/answer


        Sayan,
        Dont you have to wait for a response 183 or 200 OK before you open your 
RTP ports?

        Siddhardah


        [EMAIL PROTECTED] wrote:



                Hello All ,
                Once I send an offer in in INVITE I am supposed to be ready to 
listen to
                media.
                However If I start getting media packets before getting the 
answer , I
                do not know for sure whether the called
                Party is playing me the media or somebody else is playing me 
the media.
                What do I do in this case ? Should I be rejecting the media 
packets till
                I get an answer ?

                Also if the called party plays the media from a different IP 
address
                from the one which he provides in the the SDP (for receiving 
media)how
                can I be sure it's not some kind of attack on my UA ?
                Regards ,
                Sayan


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