On 1/3/13 9:52 AM, satya r wrote:
> Hi All ,
>
> Kindly ans my question .i have share it before 2 days but any one no
> interest to solve it. Plz clarify my doubt..
I suspect the reason you are getting no response is because your
questions are so fundamental. It is really hard to give a tutorial on
sip and VoIP in response to an email question.
I suggest you seek out a sip tutorial to get started.
I'll try to answer your questions briefly, but these answers will
probably not be useful to you until you have a basic understanding of sip.
> 1. what is q parameter.how it works ?
'q' parameters are used in a variety of places. They carry a floating
point value between zero and 1. When a group of things have q-values,
then they can be arranged into a preference order by q-value. For
instance, in REGISTER, the contacts are arranged in preference value by
q-value.
> 2 .why ACK is different transaction for successful 2xx response ?
A failing INVITE receives only one response, so there will be only one
ACK. In this case it is all treated as a single transaction.
But an INVITE can get multiple successful responses. (Due to forking of
the request by proxies.) If the ACK was part of the transaction, then
only the first successful response could be ACKed. So the ACK is treated
as a separate transaction.
This is a bit of a hack. It is often found confusing.
> 3. what is codec, what are the different video and audio codec in sip
> ,In a SIP message where i will find the codec ?
A codec is a "COer-DECoder" (of streaming media, such as audio and
video). Each codec has its own representation for the media it supports.
An simple audio session consists of two endpoints, each containing an
audio codec. Those codecs exchange messages containing encoded
representations of the audio, over a data transport connection. The two
codecs need to be matched regarding encoding and know each other's
transport type and address.
A primary purpose of SIP (Session Initiation Protoco) is to negotiate
and establish such media sessions - agreeing on how many media streams
there will be, the type of each (audio/video/...), the codec to be used
for each, and the parameters for each media stream.
Where you get your codecs depends greatly on the environment in which
you are developing.
> 4. what is media negotiation in sdp ?
The SIP INVITE exchanges SDP between the two endpoints in an
offer/answer protocol. The SDP describes the proposed media sessions (as
I just described above).
> 5.what are the basics difference between call park and call transfer ?
These are basic telephony concepts/features, not specific to sip.
They are widely implemented in sip-based telephony systems. Sip doesn't
standardize the implementation of features, but does provide mechanisms
through which they may be implemented.
Call Park is something like Call Hold. One party in an active call
detaches itself from active participation in the call, while still
leaving the call "up" as far as the other party is concerned. In the
case of Call Park, the inactive end is left in a public place where it
can be resumed by a variety of different endpoints. In Call Hold the
call can typically only be resumed by the endpoint that placed it on hold.
Call Transfer causes one end of an active call being moved from one
endpoint to another. There are various forms and user interfaces for
call transfer.
If a call is parked by one endpoint and resumed by a different endpoint
then the effect is similar to a transfer. But the user experience may be
different.
Good Luck,
Paul
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