Kedar,
Payload type field in RTP packets tell you only the codec. It does not provide
you any info regarding what the frame duration is. Payload size will give you
an idea what might be the frame duration. What if called party is using
multiple frames per packet?
-Sid
Kedar Karmarkar <[EMAIL PROTECTED]> wrote: Hello Siddhardha,
How about using the payload type field in RTP packets?
Kedar
On 5/26/06, Siddhardha Garige <[EMAIL PROTECTED]> wrote: Hi Nataraju,
This is the sample invite send out by my UA.
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.252.98:5061;branch=z9hG4bK31363c37f
Call-ID: [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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