SIP protocol - Independent of transport layer
karthikbalaguru
I find that,SIP is designed to be independent of the underlying
transport layer; it can run on Transmission Control Protocol (TCP),
User Datagram Protocol (UDP), or Stream Control Transmission
Protocol (SCTP).
Then, why is the SIP protocol referred as a " TCP/IP-based
Application Layer protocol " ? Any ideas ?
Reference -
http://en.wikipedia.org/wiki/Session_Initiation_Protocol
Thx in advans,
Karthik Balaguru
David Schwartz
wrote:
I think you're getting a bit confused between how a protocol gets
information from point A to point B and what information the protocol
gets from point A to point B. While SIP is designed to be independent
of the underlying transport layer and could theoretically get
information from point A to point B by some non-TCP/IP mechanism, what
is the actual information it's getting from point A to point B? Look
at typical SIP messages and you'll see they convey information *about*
TCP/IP protocols, addresses, and ports.
DS
karthikbalaguru
Thx for the response.
I have another query related with SIP while using Audio or Video
stream.
The voice and video stream communications in SIP applications are
carried
over another application protocol, the Real-time Transport Protocol
(RTP).
Parameters (port numbers, protocols, codecs) for these media streams
are
defined and negotiated using the Session Description Protocol (SDP)
which
is transported in the SIP packet body. And these parameters are used
to
determine the type of media(Audio/Video) and put in the corresponding
queue/buffer in the user terminal rather than parsing the entire
packet to
determine the presence or absence of audio / video.
Is the above understanding correct ? Correct me if i am wrong.
Thomas T. Veldhouse
>
> Thx for the response.
> I have another query related with SIP while using Audio or Video
> stream.
>
> The voice and video stream communications in SIP applications are
> carried
> over another application protocol, the Real-time Transport Protocol
> (RTP).
> Parameters (port numbers, protocols, codecs) for these media streams
> are
> defined and negotiated using the Session Description Protocol (SDP)
> which
> is transported in the SIP packet body. And these parameters are used
> to
> determine the type of media(Audio/Video) and put in the corresponding
> queue/buffer in the user terminal rather than parsing the entire
> packet to
> determine the presence or absence of audio / video.
>
> Is the above understanding correct ? Correct me if i am wrong.
>
You aren't asking these questions because you are looking for answers; you
already know what the answers are. What is the *real* reason that you are
asking these questions?
--
Thomas T. Veldhouse
Religion is a crutch, but that's okay... humanity is a cripple.
karthikbalaguru
> In comp.dcom.voice-over-ip karthikbalaguru <karthikbalagur...@gmail.com> wrote:
>
>
>
>
>
>
>
> > Thx for the response.
> > I have another query related with SIP while using Audio or Video
> > stream.
>
> > The voice and video stream communications in SIP applications are
> > carried
> > over another application protocol, the Real-time Transport Protocol
> > (RTP).
> > Parameters (port numbers, protocols, codecs) for these media streams
> > are
> > defined and negotiated using the Session Description Protocol (SDP)
> > which
> > is transported in the SIP packet body. And these parameters are used
> > to
> > determine the type of media(Audio/Video) and put in the corresponding
> > queue/buffer in the user terminal rather than parsing the entire
> > packet to
> > determine the presence or absence of audio / video.
>
> > Is the above understanding correct ? Correct me if i am wrong.
>
> You aren't asking these questions because you are looking for answers; you
> already know what the answers are. What is the *real* reason that you are
> asking these questions?
>
I wanted to confirm my understanding so that the design can
be visualized.
Thx,
Karthik Balaguru