A softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, typically via the internet, entirely by means of software running on a general-purpose computer system. Most landline calls are routed by purpose-built hardware, formerly using physical switchboards, but softswitches are the dominant 21st century trend.
Although the term softswitch technically refers to any such device, it is more conventionally applied to a device that handles IP-to-IP phone calls, while the phrase "access server" or "media gateway" is used to refer to devices that either originate or terminate traditional "land line" (hard wired) phone calls. In practice, such devices can often do both. As a practical distinction, a Skype-to-Skype phone call is entirely IP (internet) based, and so uses a softswitch somewhere in the middle connecting the calling party with the called party. In contrast, access servers might take a mobile call or a call originating from a traditional phone line, convert it to IP traffic, then send it over the internet to another such device, which terminates the call by reversing the process and converting the Voice over IP call back to ISDN digital or analog/PSTN format, and connecting it to a destination phone number.
A softswitch is typically used to control connections at the junction point between circuit-switched and packet-switched networks. A single device containing both the switching logic and the switched fabric can be used for this purpose; however, modern technology has led to a preference for decomposing this device into a Call Agent and a Media Gateway.
The Call Agent takes care of functions such as billing, call routing, signaling, call services and the like, supplying the functional logic to accomplish these telephony meta-tasks. A call agent may control several different media gateways in geographically dispersed areas via a TCP/IP link. It is also used to control the functions of media gateway, in order to connect with media as well as other interfaces. This procedure is utilized to keep the interfaces clear as crystal for receiving calls from any phone lines.
The Media Gateway connects different types of digital media stream together to create an end-to-end path for the media (voice and data) in the call. It may have interfaces to connect to traditional PSTN networks, such as DS1 or DS3 ports (E1 or STM1 in the case of non-US networks). It may also have interfaces to connect to ATM and IP networks, and the most modern systems will have Ethernet interfaces to connect VoIP calls. The call agent will instruct the media gateway to connect media streams between these interfaces to connect the call - all transparently to the end-users.
The softswitch generally resides in a building owned by the telephone company called a central office. The central office will have telephone trunks to carry calls to other offices owned by the telecommunication company and to other telecommunication companies via the PSTN.
Looking towards the end users from the switch, the Media Gateway may be connected to several access devices. These access devices can range from small Analog Telephone Adaptors (ATA) which provide just one RJ11 telephone jack to an Integrated Access Device (IAD) or PBX which may provide several hundred telephone connections.
Typically the larger access devices will be located in a building owned by the telecommunication company near to the customers they serve. Each end user can be connected to the IAD by a simple pair of copper wires.
The medium-sized devices and PBXs are most commonly used by business that locate them on their own premises, and single-line devices are mostly found at private residences.
At the turn of the 21st century with IP Multimedia Subsystem (or IMS), the Softswitch element is represented by the Media Gateway Controller (MGC) element, and the term "Softswitch" is rarely used in the IMS context. Rather, it is called and AGCF (Access Gateway Control Function).
Software switch, a softswitch is an API that is used to bridge a traditional PSTN and VoIP by linking PSTN to IP networks and managing traffic that contains a mixture of voice, fax, data and video. Softswitches are able to process the signaling for all types of packet protocols. Softswitch is a software-based switching platform, which is opposed to traditional hardware-based switching center technology. Softswitches also are based on open systems, another difference between them and traditional proprietary hardware switching systems.
Softswitch is also called media gateway controller, call agent and gatekeeper.
The new innovation in switching - Softswitch is far less expensive both in terms of purchase and maintenance as compared to the conventional switches used in PSTN networks. Interoperability is primarily the biggest cutting edge advantage that the service provider gets out of using a Softswitch. Softswitch is an enhancement over the existing gatekeeper technology which supported H.323, since H.323 was only restricted to LAN's the activities and capabilities of the gatekeeper were restricted to a few gateways which were managed by a single gatekeeper. As the networks became larger, there was a need for more efficient and smart solutions for managing all these services, which were answered by the softswitch. The softswitch coordinates the call control, signaling, and the other features that enables making a call across networks possible.It provides all the essential call control and service logic functions, coordinates routing of signaling messages between networks, it also provides all the administrative functions like billing statistics and providing other value added services to the users.
It serves as the intermediary between the IP and the PSTN networks. The gateway provides interoperability between the SS7 protocol used by PSTN networks and the SIP/ H.323 protocols used in VoIP.
They are used for the packetization of the circuit switched voice stream. Media Gateways are controlled by a Media Gateway Controller (Softswitch) which provides the call control and signaling functionality
It is responsible for all the services that the customer is provided with by the service provider like call voice mail, call forwarding.
Presently, the telephone companies are losing so much of their business to the VoIP counterparts, but still the speculation exists on the kind of voice quality that VoIP service providers would give to customers. The techniques used to measure the voice quality of a VoIP call are the Mean Opinion Score (MOS) and Perceptual Speech Quality Measurement (PSQM)
MOS follows the measurement techniques specified in ITU-T P.800, where different people are made to listen the voice signals and are made to rate the factors like distortion, delay, echo, noise etc on a scale of 1 to 5 where 1 is the minimum and 5 being the maximum. The mean opinion score is then calculated. A value of MOS 4 is considered as the toll quality. The conventional codec for fixed line telephones G.711 has a MOS of 4.0 at 64kbps. The modified codecs mainly used for VoIP are G.729 and G.723. G.729 is widely used for VoIP because of its low bandwidth requirements, it has a MOS value of 4.0 operates at 8kbps. The G.723 which is mainly used for Video Telephony, has a MOS 3.8 and operates at 5.3 /6.8 kbps (Table 1)
Based on ITU-T P.861 standard, this technique uses artificial speech, to provides numeric values of approximate speech intelligibility taking into account effects such as noise, coding errors, packet reordering, phase jitter, and excessive bit error rate. PSQM=0 signifies no impairment, while PSQM=6.5 indicates that the signal is totally unusable. Although PSQM values do not have any direct correlation with MOS values, but roughly PSQM value of 0 corresponds to MOS of 5 and PSQM value of 6.5 to MOS of 1.
The other factors affecting the voice quality are :
Voice transmission over wireless brings along with a it a big problem of packet delay or latency. The factors that add to the delay are the Propagation delay, the serialization delay, channel coding delay at the physical layer and the Medium access delay at the MAC layer. Similarly at the Network layer the Forwarding and the Queuing delays are encountered, and at the application layer Packetization / depacketization, delay, Algorithmic delay and look-ahead delay, decoding delay are inherently caused. Studies have proved that packet delay of 100ms doesn't do any harm, but if the delay increases to 150ms the voice signal is unusable. The service providers have to ensure that the delay caused does not exceed 100ms by any ways.
Packet Loss/ Dropped Packets
Packet loss does excessive damage to the voice signal, as retransmission cannot be considered as an option while transmitting voice.Loss of voiced frames at unvoiced/voiced transition causes significant degradation of the signal. Advanced error detection and correction algorithms are used to fill the gaps created by the dropped packets. A sample of the speaker's voice is stored and is used to create a new sound from an algorithm which tries to approximate the contents of the dropped packets or lost packets.
When the transmission times of the arriving packets varies as a result of different queuing times or different routes it is referred to as Jitter. Jitter can be taken care of by using an adaptive jitter buffer which adapts itself according to the delay encountered over the networks, to provide a smooth voice stream at the output.
Multiprotocol Label Switching (MPLS) is a mechanism in high-performance telecommunications networks that directs data from one network node to the next based on short path labels rather than long network addresses, avoiding complex lookups in a routing table. The labels identify virtual links (paths) between distant nodes rather than endpoints. MPLS can encapsulate packets of various network protocols. MPLS supports a range of access technologies, including T1/E1, ATM, Frame Relay, and DSL.
MPLS is a highly scalable, protocol agnostic, data-carrying mechanism. In an MPLS network, data packets are assigned labels. Packet-forwarding decisions are made solely on the contents of this label, without the need to examine the packet itself. This allows one to create end-to-end circuits across any type of transport medium, using any protocol. The primary benefit is to eliminate dependence on a particular OSI model data link layer technology, such as Asynchronous Transfer Mode (ATM), Frame Relay, Synchronous Optical Networking (SONET) or Ethernet, and eliminate the need for multiple layer-2 networks to satisfy different types of traffic. MPLS belongs to the family of packet-switched networks.
MPLS operates at a layer that is generally considered to lie between traditional definitions of layer 2 (data link layer) and layer 3 (network layer), and thus is often referred to as a "layer 2.5" protocol. It was designed to provide a unified data-carrying service for both circuit-based clients and packet-switching clients which provide a datagram service model. It can be used to carry many different kinds of traffic, including IP packets, as well as native ATM, SONET, and Ethernet frames.
A number of different technologies were previously deployed with essentially identical goals, such as Frame Relay and ATM. MPLS technologies have evolved with the strengths and weaknesses of ATM in mind. Many network engineers agree that ATM should be replaced with a protocol that requires less overhead, while providing connection-oriented services for variable-length frames. MPLS is currently replacing some of these technologies in the marketplace. It is highly possible that MPLS will completely replace these technologies in the future, thus aligning these technologies with current and future technology needs.
In particular, MPLS dispenses with the cell-switching and signaling-protocol baggage of ATM. MPLS recognizes that small ATM cells are not needed in the core of modern networks, since modern optical networks (as of 2008) are so fast (at 40 Gbit/s and beyond) that even full-length 1500 byte packets do not incur significant real-time queuing delays (the need to reduce such delays — e.g., to support voice traffic — was the motivation for the cell nature of ATM).
While the traffic management benefits of migrating to MPLS are quite valuable (better reliability, increased performance), there is a significant loss of visibility and access into the MPLS cloud for IT departments.
MPLS works by prefixing packets with an MPLS header, containing one or more labels. This is called a label stack. Each label stack entry contains four fields:
- A 20-bit label value.
- a 3-bit Traffic Class field for QoS (quality of service) priority (experimental) and ECN (Explicit Congestion Notification).
- a 1-bit bottom of stack flag. If this is set, it signifies that the current label is the last in the stack.
- an 8-bit TTL (time to live) field.
These MPLS-labeled packets are switched after a label lookup/switch instead of a lookup into the IP table. As mentioned above, when MPLS was conceived, label lookup and label switching were faster than a routing table or RIB (Routing Information Base) lookup because they could take place directly within the switched fabric and not the CPU.
Routers that perform routing based only on the label are called label switch routers (LSRs). The entry and exit points of an MPLS network are called label edge routers (LERs), which, respectively, push an MPLS label onto an incoming packet and pop it off the outgoing packet. Alternatively, under penultimate hop popping this function may instead be performed by the LSR directly connected to the LER.
Labels are distributed between LERs and LSRs using the Label Distribution Protocol (LDP). LSRs in an MPLS network regularly exchange label and reachability information with each other using standardized procedures in order to build a complete picture of the network they can then use to forward packets. Label-switched paths (LSPs) are established by the network operator for a variety of purposes, such as to create network-based IP virtual private networks or to route traffic along specified paths through the network. In many respects, LSPs are not different from permanent virtual circuits (PVCs) in ATM or Frame Relay networks, except that they are not dependent on a particular layer-2 technology.
In the specific context of an MPLS-based virtual private network (VPN), LERs that function as ingress and/or egress routers to the VPN are often called PE (Provider Edge) routers. Devices that function only as transit routers are similarly called P (Provider) routers. See RFC 4364. The job of a P router is significantly easier than that of a PE router, so they can be less complex and may be more dependable because of this.
When an unlabeled packet enters the ingress router and needs to be passed on to an MPLS tunnel, the router first determines the forwarding equivalence class (FEC) the packet should be in, and then inserts one or more labels in the packet's newly created MPLS header. The packet is then passed on to the next hop router for this tunnel.
When a labeled packet is received by an MPLS router, the topmost label is examined. Based on the contents of the label a swap, push (impose) or pop (dispose) operation can be performed on the packet's label stack. Routers can have prebuilt lookup tables that tell them which kind of operation to do based on the topmost label of the incoming packet so they can process the packet very quickly.
In a swap operation the label is swapped with a new label, and the packet is forwarded along the path associated with the new label.
In a push operation a new label is pushed on top of the existing label, effectively "encapsulating" the packet in another layer of MPLS. This allows hierarchical routing of MPLS packets. Notably, this is used by MPLS VPNs.
In a pop operation the label is removed from the packet, which may reveal an inner label below. This process is called "decapsulation". If the popped label was the last on the label stack, the packet "leaves" the MPLS tunnel. This is usually done by the egress router, but see Penultimate Hop Popping (PHP) below.
During these operations, the contents of the packet below the MPLS Label stack are not examined. Indeed transit routers typically need only to examine the topmost label on the stack. The forwarding of the packet is done based on the contents of the labels, which allows "protocol-independent packet forwarding" that does not need to look at a protocol-dependent routing table and avoids the expensive IP longest prefix match at each hop.
At the egress router, when the last label has been popped, only the payload remains. This can be an IP packet, or any of a number of other kinds of payload packet. The egress router must therefore have routing information for the packet's payload, since it must forward it without the help of label lookup tables. An MPLS transit router has no such requirement.
In some special cases, the last label can also be popped off at the penultimate hop (the hop before the egress router). This is called penultimate hop popping (PHP). This may be interesting in cases where the egress router has lots of packets leaving MPLS tunnels, and thus spends inordinate amounts of CPU time on this. By using PHP, transit routers connected directly to this egress router effectively offload it, by popping the last label themselves.
MPLS can make use of existing ATM network or Frame Relay infrastructure, as its labeled flows can be mapped to ATM or Frame Relay virtual-circuit identifiers, and vice versa.
Installing and removing paths
There are two standardized protocols for managing MPLS paths: the Label Distribution Protocol (LDP) and RSVP-TE, an extension of the Resource Reservation Protocol (RSVP) for traffic engineering. Furthermore, there exist extensions of the Border Gateway Protocol (BGP) that can be used to manage an MPLS path.
An MPLS header does not identify the type of data carried inside the MPLS path. If one wants to carry two different types of traffic between the same two routers, with different treatment by the core routers for each type, one has to establish a separate MPLS path for each type of traffic.
Multicast was for the most part an after-thought in MPLS design. It was introduced by point-to-multipoint RSVP-TE. It was driven by service provider requirements to transport broadband video over MPLS. Since the inception of RFC 4875 there has been tremendous surge in interest and deployment of MPLS multicast and this has led to several new developments both in the IETF and in shipping products.
MPLS and WiMAX
A WiMAX base station provides wireless connectivity at the physical layer. In particular, WiMAX provides last mile connectivity to the end user. It can also be used to provide point-to-point (PTP) backhaul links.
From this perspective, anything above the physical layer can run transparently on WiMAX. While this is so, WiMAX has defined Convergence Sublayers (CS) at its interface which can then be mapped correctly to 802.16 MAC layer before the packets are sent on the wireless channel. The supported CS Specifications include ATM and Packet (IPv4, IPv6, Ethernet, 802.1Q-VLAN) CS. So where does MPLS fit in, if at all?
MPLS is a technology that sits between Layer 2 and Layer 3. It can be seen to be outside the scope of a WiMAX base station and certainly outside the scope of the WiMAX standards. This is where we, as engineers, have to look at the whole thing from a deployment and operational angle.
Firstly, operators want MPLS because of the many advantages it offers. It leverages on both IP and Ethernet, technologies which are cheap and ubiquitous. It offers QoS. It provides multipoint connectivity but in a simpler way than IP. Its faster to switch at Layer 2 using labels than perform routing decisions at Layer 3. The attractiveness about MPLS in the coming years is that it is set to enable the move towards all-IP transport networks. When IP replaces TDM and ATM architectures, MPLS is set to play a major role.
So operators are interested in MPLS. Before they install new devices into their network, they want to know if it supports MPLS. The problem is that there is a clear distinction between core networks and access networks. MPLS is usually limited to the core. However, there has been significant push towards bringing MPLS to the access networks. This enables end-to-end traffic engineering, right up to the WiMAX base station. 3ROAM offers such a base station with MPLS built-in. Likewise, New Edge Networks is another company that is taking MPLS beyond the core to edge networks.
What if the WiMAX base station did not support MPLS? In this case, an MPLS-enabled network would terminate at an MPLS edge router (ingress or egress). This router would then be co-located and connected to the WiMAX base station. The problem for the operator in this situation would be to have a router for every base station. This is simply not cost-effective.
In general, WiMAX base stations operate in bridge mode (Layer 2) or routing mode (Layer 3). If a base station has to be MPLS-enabled, it has to work in Layer 3 mode. In other words, the base station doubles as an ingress/egress router. It does more than simply provide wireless connectivity.
Sprint has in its long-term roadmap this architecture in mind for backhauling of its WiMAX network. Sprint’s WiMAX base stations would be MPLS-enabled and the backhaul between such a base station and its ASN Gateway would be IP over MPLS. One of Sprint’s providers for its WiMAX backhaul is Ciena which uses PBB-TE. This may very well carry IP over MPLS right up to the base station. The backhaul itself is wireless with equipment supplied by DragonWave.