QoS: QoS is a collection of technologies which allows applications to request and receive predictable service levels in terms of data throughput capacity (bandwidth), latency variations (jitter), and delay.
Given below are the QoS considerations and what it means:
1. Classification: Identifies and marks flow and provides priority to certain flows
2. Congestion management: It is a mechanism to handle traffic overflow using a queuing algorithm
3. Link-efficiency mechanisms: Reduce latency and jitter for network traffic on low-speed links
4. Traffic shaping and policing: Avoids congestion by policing ingress and egress flows
The default mechanism on most interfaces is First-In First-Out (FIFO). Some traffic types have more demanding delay/jitter requirements. Thus, one of the following alternative queuing mechanisms may be configured or is enabled by default:
Priority Queueing (PQ)
Custom Queueing (CQ)
Weighted Fair Queueing (WFQ)
Class-Based Weighted Fair Queueing (CBWFQ)
Low Latency Queueing (LLQ), which is in fact CBWFQ with a Priority Queue (PQ) (known as PQCBWFQ)
Low-latency queuing (LLQ) adds a strict priority queue (PQ) to CBWFQ (Class Based Weighted Fair Queuing). It allows delay-sensitive traffic such as voice to be sent first, before other queues are serviced. Unlike priority queuing, LLQ provides for a maximum threshold on the PQ to prevent lower priority traffic from being starved by the PQ. LLQ is most widely used QoS method for many VoIP networks.
Cisco routers and switches use Weighted Fair queuing for interfaces below 2 Mbps by default.
The following are the Cisco recommended best practices for IPT (IP Telephony):
1. Use separate VLANs and IP subnets for IP phones and data to provide ease of QoS configuration.
2. Use private IP addresses for IP phone subnets for better security to voice devices.
3. Place CallManager and Unity servers on filtered VLAN/IP subnets in the server access in the data center.
4. Use IEEE 802.1Q trunking and 802.1P to allow for prioritisation at Layer 2.
5. Extend QoS trust boundaries to voice devices but not to PCs and other data devices.
6. In the access layer, use multiple egress queues to provide priority queuing of RTP voice streams.
7. Use DSCP for classification and marking.
8. Use Low Latency Queuing (LLQ) on WAN links. The Low Latency Queuing feature enables low latency behaviour for a traffic class for both VIP and non-VIP platforms. Low Latency Queuing allows delay-sensitive data such as voice to be prioritised and sent first, giving delay-sensitive data preferential treatment over other traffic.
9. Use LFI (link fragmentation and interleaving) on WAN links less than 768 kbps.
10. Use CAC to avoid oversubscription of circuits.
11. IPT voice packets should be marked with a DSCP of EF (IP precedence 5). Signalling packets should be marked with AF31 (IP precedence 3).
Propagation delay: Fixed delay of 6 ms per km.
Serialization delay: Frame length/bit rate. A faster link and smaller packets help reduce.
Processing delay: Depends on codec used: coding, compression, and packetization.
Queuing delay Variable packet sizes and number of packets results in variable delay.
Jitter Caused by variable delay. Use dejitter buffers to make delay constant; design as much as possible for an uncongested network.
|IPT Functional Area||Description|
|Service applications||Unity, IVR, TAPI interface|
|Call processing||Cisco CUCM|
|Client endpoints||IP phones, digital and analog gateways|
|Voice-enabled infrastructure.||Layer 2 and Layer 3 switches and routers|
A number of different protocols are used in a VoIP environment for call control, device provisioning, and addressing. Figure below shows those protocols relevant to VoIP traffic:
As per Cisco guidelines while designing a VOIP network, the total bandwidth for voice, data, and video should not exceed 75 percent of the provisioned link capacity during peak hours. The remaining bandwidth is used by routing, multicast, and management protocols.
The bandwidth requirement for VoIP call may be calculated as below:
1. G.711 codec uses 64 kbps bit rate
2. IP/UDP/RTP header length 40 bytes
3. Default voice payload = 160 bytes * (8 bits/bytes) = 1280 bits
4. WAN header = 6 bytes
5. Voice packet size = 6 bytes + 40 bytes + 160 bytes = 206 bytes * (8 bits/byte) = 1648 bits
6. PPS = 64 kbps / 1280 bits = 64,000/1280 = 50 pps
7. BW per call = 1648 (bits/packet) * 50 (pps) = 82400 bps = 82.4 kbps
1. Propagation delay 6 ms per km.
2. Serialization delay : Frame length/bit rate. Serialization delay is inversely proportional to the link bandwidth. Higher the link speed, lower the delay.
3. Processing delay : Depends on codec used : coding, compression, and packetization.
1. Queuing delay : Variable packet sizes and number of packets.
2. Jitter : Caused by variable delay. Use dejitter buffers to make delay constant.
Q.SIG Q.SIG is the preferred signalling protocol used between PBX switches.
SS7 allows voice network calls to be routed and controlled by call control centres. SS7 is used between PSTN switches. SS7 implements call set-up, routing, and control. With SS7, telephone companies can implement services such as caller ID, toll-free numbers, call forwarding, and so on.
If a group of users makes/receives 100 calls in the average busiest hour and each call lasts and average of 2 minutes, the Erlangs are calculated as follows:
100 calls per hour * 2 minutes per call = 200 minutes per hour
Traffic volume = (200 minutes per hour) / (60 minutes per hour) = 3.33 Erlangs
The following components belong to Cisco Voice Enabled Infrastructure
1. QoS Enabled Switches (L2, L3 layers)
The following are the design goals for IPT network
1. To provide cost-effective VoIP facility
2. To provide the reliability and high availability traditionally associated with traditional voice technologies
3. To provide significantly lower cost of ownership than traditional telephony
4. To offer more number of features such as remote worker, mobility, etc that are not available with traditional telephony.
5. To provide new applications such as tele presence, IVR, contact centres, etc.
6. To facilitate consolidation of data and telephony infrastructures
7. To ensure backward compatibility with traditional telephony.
Given below are the typical deployment models for Cisco IPT call processing:
The Single-Site model is designed to be locally managed and administered. It can operate on a wired or wireless LAN. Local and long distance calling is achieved through gateway connectivity with the PSTN by various combinations of T1/E1 CAS and PRI.
Multisite WAN with centralized call processing:
The Multisite Centralized Call Processing model is suitable for businesses such as banks, which include a corporate headquarters and branch offices at remote locations.
Multisite WAN with distributed call processing:
Each site in the Multisite Distributed Call Processing model can operate on a wired or wireless LAN. The inter-site WAN connection can be frame relay, MPLS, or site-to-site VPN. Each branch site can operate on a wired or wireless LAN.
Local calling is achieved through gateway connectivity at each site. Long distance calling for each site uses the WAN for on-net calling. Off-net long distance traffic is backhauled over the WAN to one or more drop-off gateways.
Cisco CallManager Express deployment:
The model for a Multisite deployment with distributed call processing consists of multiple independent sites, each with its own call processing agent cluster connected to an IP WAN that carries voice traffic between the distributed sites.
The following are true within media service framework
1. Access services include identity of end devices, and mobility.
2. Transport services include QoS for reliable packet delivery.
3. Bridging services include transcoding, and recording services of media streams.
4. Storage services provide capture and storage of media and content management.
5. Session control services provide session signalling and control.
Media Gateway Control Protocol (MGCP), also known as H.248, is a standard protocol for handling the signalling and session management needed during a multimedia conference. In MGCP networks, endpoints cannot function without communication and control from the call agent. Cisco CUCM is an example of call agent.
LFI and cRTP helps in reducing the serialization delay on slow-speed WAN circuits. LLQ will help when there is congestion in the circuit.
ISR routers 800, 1900,2900,3900 deliver security, voice, and IP services with support for rich-media applications suck as video and on-demand services.
4000 delivers converged routing platforms for branch networks.
ASR 1000 delivers high performance routing for network edge and data center.
ASR 9000 provides high availability and scalability for SP Edge
4900 provides speeds up to 2Gbps, and used for LAN switching in enterprise access and distribution layers in campus.
Cisco recommends G.729 codec for WAN links because of its lower bandwidth requirements and better MOS compared with G.723 and others.
1. G.711 has a Mean Opinion Score of 4.3-4.7 and uses 80 kpbs (if you send 50 packets/second with 20 ms of RTP payload per packet) or 74.7 kbps (@ 30 ms, meaning 33.3 packets/second).
2. G.729 (NOT G.729a) has a MOS of 3.9-4.2 and uses 24 kbps @ 20 ms or 18.7 kbps @ 30 ms.
3. G.729a has a MOS of 3.7-4.2 and uses 24 kbps @ 20 ms or 18.7 kbps @ 30 ms.
4. G.723 has a MOS of 3.8-4.0 and uses 17.1 kbps @ 30 ms.
MOS of 5.0 is the maximum score. All of the above numbers are in EACH direction, so total bandwidth is double the above figures.
Call admission control (CAC) is used to regulate traffic volume in voice communications, particularly in wireless mobile networks and in VoIP networks. Call admission control is also used to maintain a certain level of audio quality in voice communications networks, or a certain level of performance in Internet nodes and servers where VoIP traffic exists.
Voice gateways are devices that communicate with other voice gateways, gatekeepers, their respective endpoints, and call control agents, such as Cisco Unified Communications Manager or a PBX via voice signalling and media protocols which include MGCP, H.323, SIP, SCCP and RTP. In VoIP networks, the primary purpose of voice gateways is to provide an interface between the VoIP network and the Public Switched Telephone Network (PSTN).
A typical IP telephone network is shown in the figure below:
As shown in the diagram above, the voice gateway is connected to both the VoIP network and the PSTN. The gateway interfaces with the IP network and the PSTN and supports IP signaling control protocols used in Voice over IP, and Time Division Multiplexing (TDM) control protocols used on the PSTN.
Services provided by IP telephony network
a. Transport Services ................ Provide QoS for reliable packet delivery
b. Bridging Services .................. Transcoding and Bridging
c. Storage Services ..................... Capture and Storage of media services
d. Session Control Services ..... Provide session signalling & control and gateway services
e. Access Services ....................... Provide identity of end devices, mobility, and location services
TelePresence requires 4 -12 Mbps bandwidth for high definition video. Video surveillance requires 3 - 4 Mbps per camera and the bandwidth varies depending the video quality required.
Basically, a voice gateway contains an IP interface and a legacy telephone interface. This allows the voice gateway to allow for communication between the IP network and the telephone network by translating between the different transmission formats and protocols used on the two networks.
Cisco voice gateways support analog or digital telephony ports, and sometimes both port types, in addition to IP interfaces.
Supported analog ports include the following:
a. Foreign Exchange Office
b. Ear and Mouth
c. Foreign Exchange Station
Foreign Exchange Office (FXO) ports are used to connect to the PSTN or to a PBX. These are the ports on subscriber devices, such as analog telephones, fax machines and modems that connect to the PSTN via an FXS port. An Ear and Mouth (E&M) port is used to interconnect PBX devices using dedicated circuits from the PSTN.
A Foreign Exchange Station (FXS) port is used to connect analog devices, such as analog telephones, fax machines, and modems. FXS ports are the ports that subscribers plug a telephone, fax machine or modem into. FXS ports provide telephony services for these devices. FXO, E&M, and FXS ports will be described in detail later in this chapter in the section pertaining to voice ports.
Cisco voice gateways also support digital ports and interfaces, which include the following:
1. T1 Primary Rate Interface
2. T1 Channel Associated Signalling
3. E1 Primary Rate Interface
4. E1 R2
5. ISDN Basic Rate Interface