With the increase in subscriber base and an increasing need to offer a variety of services to the subscribers, operators are pushing for more lucrative and revenue generating services to end-users. The foremost demand is to introduce IP Multimedia services (IMS) to wireless subscribers, with an eventual goal of all voice over packet switch domain.
IMS allows users to show their availability, status, location to other users in a social network. On the basis of that knowledge, users will initiate more voice and SMS within the operator's network.This will also enable interactive communication sessions among users: gaming - video conferencing - chat sessions – VoIP - Messaging and a host of other applications.
The goal of introducing the IP multimedia Core Network (IM CN) subsystem into the Third Generation Partnership Project (3GPP) is to enable the convergence of, and access to, voice, video, messaging, data and web-based technologies for the wireless user, and combine the growth of the Internet with the growth in mobile communications.
3GPP and SIP
There were several competing approaches in the market with CS-Fallback being the first go option due to existing CS switched domains, proposal by Kineto wireless for VoLGA architecture using generic access network (GAN). OneVoice is an initiative from several vendors to converge on a multimedia delivery due o these many alternate approaches and convereged on the IMS architecture. IMS is based on Session Initiation Protocol (SIP) standard for interfaces to IM CN subsystem. The decision to use SIP as a call control method helps to completely decouple the services provided by the network that the user is currently in. The limitations of the visited network will not have an impact on the services that are available to the user. This helps the network operators to deploy new services much faster, since they will no longer need to wait for support from their network partners. It also allow service providers to deploy proprietary service nodes in their network. This allows innovative services which would otherwise not be possible, giving the truly creative service providers an edge in attracting and retaining customers.
There are two reasons this technique is possible:
- By not relying on the visited network to provide service triggering, the service provider is allowed to hide trade secrets which may be relevant to these new services.
- The service provider does not need to be constrained to the type of information that can be expressed by the standardized service API
SIP is a multimedia signaling protocol used to establish, terminate and modify communications sessions in an IP network. SIP relies on the underlying mechanism for providing mobility. Since mobility is provided by the access mechanism in General Packet Radio Service (GPRS), SIP is ideally suited to managing multimedia sessions for a mobile user.
Other advantages are:
· The identity of the cell through which the 3GPP User Equipment (UE) is accessing the IM CN Subsystem (Cell-ID) is transmitted during Registration or Session initiation. This is used by the home network to provide localized services such as directory assistance, information on offers at nearby shopping malls, weather forecasts, event notifications, etc.
· Users can give information to allow other users to know the status, availability and location of this mobile.
· The SIP Registration procedure makes the client’s location available to others and the client can receive incoming SIP INVITE messages (to join multimedia conferences) using the home network.
· The UE may send out a SIP INVITE message and invite other users to join a multimedia conference hosted by the 3GPP UE.
· SIP supports compression algorithms which help in efficient utilization of precious radio resources.
· SIP brings along with it, a host of useful applications to end-users as well as value added services such as presentation of remote party identification, blocking calls based on remote party identity, anonymous session establishments etc.
· Allows a seamless integration of “smart” wireless terminals into the exploding information/communication IP networks
SIP uses IPv6 functional entities in a 3GPP environment. 3GPP requires multiple outbound SIP proxies. The following entities are introduced in 3GPP:
- P-CSCF (Proxy Call Station Control Function) acting as Proxy (as defined in SIP RFC2543)
- S-CSCF (Serving CSCF) acting as UAS (User Agent Server) or Registrar or UA (User Agent) (terms as defined in SIP RFC2543)
- I-CSCF (Interrogating CSCF) used to discover S-CSCF and to hide the network topology of S-CSCF. This is the contact point within an operator’s network for all connections destined to a subscriber of that network operator.
Let us illustrate this with an example. I want to host a multimedia session on my 3GPP terminal, and I wish to invite my friends to join me. The typical flow of events would be as follows.
- GPRS Call Flow
· It emits a GPRS-attach message to the Serving GPRS Support Node (SGSN).
· Subsequently my terminal sends out a Activate-Packet Data Protocol (PDP) Context message to SGSN.
· SGSN establishes a GPRS Tunneling Protocol (GTP) tunnel towards the Gateway GPRS Support Node (GGSN) by sending Create-PDP-Context request message.
· GGSN sets up resources for my terminal and may allocate an IP address, if the Create-PDP-Context did not include an IP address.
· If successful, GGSN sends back a Create-PDP-Context response to SGSN, which in turn sends a Activate-PDP-context response message to my terminal.
At this time, my terminal is ready to send and receive data to/from the GPRS network. In an LTE EPC domain, the default bearer is established upon initial attach and the network triggers 2 dedicated bearers, one with QCI 5 (recommended value) for IMS signaling and another with QCI 1 (recommended) for voice. Upon successful establishment of these bearers, the UE registers with the IMS subsystem.
- Initiating SIP
To initiate SIP, my terminal needs to know the P-CSCF address for Registration. If my 3GPP terminal does not have DHCP capability, P-CSCF address(es) is transferred to my terminal within the PDP Context Activation Response message. My terminal would indicate that it requests a P-CSCF IP address(es) and GGSN returns the P-CSCF address which is either configured statically or obtained from radius server.
Let us assume that my terminal is DHCP/DNS capable and also supports SIP extensions (specified in draft-ietf-sip-dhcp-03.txt). It will then use DHCP to obtain a P-CSCF address from the GGSN, which also doubles as a DHCPv6 server. In addition to the P-CSCF address, my terminal will also obtain the domain name and the IP address of DNS servers. It will then use this DNS server to resolve a list of P-CSCF IP addresses from which one address is selected.
- SIP Registration
My terminal has to be registered with the Home CSCF for association between SIP Uniform Resource Locators (URLs) and various contacts. The figure below describes the callflow for a Registration transaction which includes a I-CSCF in the home network. Other SIP transactions follow similar lines.
An Example SIP Signalling Scenario where my terminal is outside of my Home network:
The terminal sends a SIP Register request to P-CSCF. The P-CSCF will perform two actions, binding and forwarding.
The binding is between the user’s SIP address (user1@home1.net) and the host (terminal) address ([5555::aaa:bbb:ccc:ddd]) which was acquired during PDP context activation process. Then P-CSCF performs a DNS Query to locate the I-CSCF in the home network. The look up in the DNS is based on the address specified in the Request URL. The P-CSCF forwards the Register request to I-CSCF.
The I-CSCF forwards the SIP REGISTER to the S-CSCF selected. The S-CSCF updates HSS about S-CSCF name and location for this subscriber. The S-CSCF downloads the subscriber profile from the HSS. The S-CSCF sends response indicating successful registration which will traverse the path that the REGISTER request took as described in the Via list. (term Via as defined in SIP RFC 2543)
The P-CSCF sends a first NOTIFY request to my terminal in order to confirm the successful subscription. A new Call-ID is generated. The terminal stores this Call-ID, CSeq, etc. and generates a 200 OK response to the NOTIFY.
- Inviting Others to the Conference
The Session establishment follows the SIP specification. The terminal determines the complete set of CODECS that it is capable of supporting for this session. It builds a Session Description Protocol (SDP) packet containing bandwidth requirements and characteristics of each, and assigns local port numbers for each possible media flow.
The terminal sends out an invitation containing the initial SDP to P-CSCF which is forwarded to S-CSCF. S-CSCF in turn will send out the invitation to the destination IP address (already registered to with S-CSCF). Upon receipt of Session Progress indication, QoS resources are authorized on P-CSCF and Resource Reservation is done at my terminal. Upon receipt of the final Acknowledgement message, the two parties are ready to go.
The Session Announcement Protocol (SAP) and SDP support the establishment of multi-party conferencing sessions. SAP defines the procedure for advertising conferencing sessions by periodically multicasting information about active sessions. SDP supports the description of multimedia sessions, including the specification of preferred media types and scheduling information. The SAP and SDP combine to provide a means of advertising sessions so that interested parties can join.
The media stream from my terminal is multi-casted to all interested participants via the S-CSCF. Thus my 3GPP terminal is now playing a movie and my friends are watching over their 3G phone!!!
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