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First of all now days packet switching is used for data transfer.
There are numerous advantage of packet switching over circuit swiching.

In circuit switching a dedicated link is provided for the data transmission.
In packet switching different packets of data is routed through different paths. Advantage of doing this is all the paths are available to use for the transmission. By packet switching, capacity of the medium to transmit data is drastically improved than by using circuit switching.

TCP/IP are set of rules that are used to transfer data.
I have said above in packet switching data is first broken down to fixed number of packets and then send through medium now how the data is broken or how it chose different paths and at the end how it is assembled again in the original form is done by the protocols of TCP/IP.

Hi,

I'll try my best awnsering your questions.

1. What is data path and signal path?

Data path refers to the logic/physical route that follow all the information that users send over a telco network. Normally, the data path flow is set between final users (one person sending an e-mail to another, you calling your girl friend) but this cannot be generalized specially if we consider M2M communications.
Signal path refers to the logic/physical route that follows all the information between the user and network elements and between network elements in order to coordinate the network operation. Telecommunication networks use signaling to do things like establishment and release of connections, network monitor…
Simple Example:
Signaling A to B: I want to talk to you, are you ready?
Signaling B to A: Yes, I’m ready, are you ready?
Signaling A to B: Me to, do you agree to talk in English?
Signaling B to A: Prefect to me.
Data A to B: What do you think about UK position in the European Union?
Data B to A:…

2. Relation among CS call, PS call and VoIP call ?

In a CS call (like in traditional telephonic networks) each call requires reserved resources between both sizes of the communication (dedicated channel). This leads to bad network usage since resources are not optimally used but easily ensures service quality without fancy operation strategies.
PS is the usual way that the Internet (TCP/IP architectures) uses in order to send information. When you send an e-mail through the Internet the info f that e-mail is divided in smaller packets (information pieces). Since no channel reservation is made, each packet of your e-mail can follow a different route through the Internet (for example, one packet can go from London to Rome through Paris and another through Madrid). This strategy leads to better resource utilization but makes more difficult to achieve some quality levels.
VoIP is a term that refers to the protocols used to send voice digitally over an IP network like Internet. Basically, a CS call is a call over a traditional circuit switching networks like 2G mobile networks or fixed telephonic networks. A VoIP call uses some protocols (audio codecs, signaling protocols…) to achieve over a packet switching network (that are more efficient than CS networks) the same voice quality than in a traditional CS networks. Without VoIP a PS call would not have a proper quality do, among other problems, the delay problems over PS networks.

3. How does TCP/IP comes into picture while doing a PS call using GPRS?

I think GPRS is a GSM service to send data over PS networks. In other words, on 2G networks voice calls go over a CS (dedicated resources) and data transmissions go through a PS network (TCP/IP). One way to send voice over 2G networks is using VoIP. Anyway, on a GPRS network TCP/IP are transport and network protocols used to send information through the transport network (users are connected to access networks that are connected through a transport network). Since GSM uses a specific protocol stack over the radio access, another protocol is used to transparently transmit access network packets on a TCP/IP network (GTP).

4. Difference between PS call and web browsing using GPRS ( flow path difference)

PS call over GPRS is possible using VoIP. In order to perform a VoIP communications you need a terminal with proper audio codecs to digitalize/ compress voice before sending it to lower protocols (TCP/UDP - IP). In addition to voice codecs the terminal must support some signaling functions that emulate, among others, the calling setup/release process of traditional CS telephonic networks. These elements (audio codecs and signaling protocols) work over TCP or UDP and IP at the protocol stack. This means that the codified audio result of applying a codec audio is passed to the TCP/UDP protocol in order to ensure that the message is entirely sended to the other size. On a similar way, IP split the information in smaller packets in order to send them more efficiently over the network.

TCP and UDP are two different transport protocols. They are used for the same purpose (send information between end terminals) but they do it in different ways. I’ll not go in detail but while TCP is used to send signaling, UDP is used to send voice. Why? Well, the main reason is that voice calls require low latency but are tolerant to some losses. In such a scenario UDP, a transport protocol that doesn’t retransmit loosed information, is quite good. On the other side, TCP ensures that all the info arrives by retransmitting and acknowledging all the packets. This is not that good for voice call info but its perfect for signaling.

Sorry for my basic English. I hope the info will be useful to you.

Happy new year

Hi,

The difference between signaling and data...

Best example is taking your telephone network. The twisted pair cable is present. In twisted pair one line is for signal. another one is for you voice/packet data. When you pickup the receiver you will get the dial tone. Actually, the signal inform to the circuit switch device, user's going to dial.
When you are dialing you hear some dial sound , actually its establishing connection based on your dial number. Once you done dialing. the signal reach the destination number and inform to that device and you will hear the ringing sound.
Once the destination party pickup the call. The call is established and you ready to speak. So all the signals are going and come by one line. The data is in another line. Now a days in telephone system they are using SS7 signaling protocol...

For CS and PS refer in google.(Please understand what is CS and PS then u can able to understand my below notes... :) )

For VoIP call. First they must establish the Session (in telecom we using CS so signal establish based on dial number. but in PS is not happen like that so session must need). Session is just establish the connection between two party. The session will happen with the help of SIP signaling protocol (SS7 for CS network.)

Once session will over those party will speak . The audio information going with the help of RTP protocol (in CS network, they doing analog to digital and transmit using mux and other operations. but ps is packet based so its need some protocol to reach the destination so here using RTP).

Regards,
Karthikeyan S