QoS

General

Quality of Service (QoS) refers to control mechanisms in data networks that try to ensure a certain level of performance to a data flow in accordance with requests from the application program. Data equipment tries to ensure the requested level of performance among others by performing

• Call Admission Control,
• Classification of Packets,
• Class based Packet processing
  • Queuing and Scheduling,
  • Traffic Shaping.
  • Congestion Avoidance (random early detect, RED)

It is the target of Quality of Service mechanisms to minimize

• packet drops
• delay
• jitter (variable delay)

especially for high priority traffic flows.

Call Admission Control

Mechanisms like Queuing and Scheduling help to priviledge high prority traffic (e.g. real-time traffic) over other data traffic. However, if the sum of all high priority flows exceed the capabilites of network links or elements, those techniques will fail. An additional high priority flow might degrade the quality of all existing flows.

Call Admissin Control makes sure that existing flows will keep their quality by rejecting any additional high priority flow. In the terms of telephony, the call is rejected and will either receive a busy signal or is rerouted.

Centralized Static Call Admission Control

Siemens HiPath 8000 has implemented a centralized static Call Admission Control which works in hub-and-spoke or star topologies. Topologies are abstracted to zones with unlimited bandwidth ressources which are interconnected by bottlenecks. If phone A_1 resides in zone_A only, if the bandwidth needed does not exceedand phone B_1 resides in zone_B and those zones are interconnected with a bottleneck link of 150 kbps, the call of 100 kbps is admitted, if no other call is using the bottleneck [link. Now, a second phone A_2 in zone_A wants to start an other 100 kbps call to phone B_2 in zone_B. Since the needed bandwidth would exceed the left bandwidth ressources of 50 kbps, the call will be rejected, preserving the call quality of the first call.

In the case of the HiPath 8000 solution, telephones are classified to be members of certain zones based on IP-addresses or telephone numbers.

Cisco CallManager makes use of the very same centralized static Call Admission Control and also is limited to hub-and-spoke or star topologies.

Network based dynamic Call Admission Control

A standard way of performing Call Admission Control is Ressource ReserVation Protocol (RSVP), i.e. the "IntServ Model". With RSVP, each end system that seeks to send a data flow will send an RSVP PATH message to its communication partner. Network elements along the path that are RSVP aware will add route records to the RSVP PATH messages; others will route the packets like normal IP packets. The communication parter will respond to the RSVP message with a RESV packet and will send it back along the recorded path. RSVP aware network elements now will perform the reservation, if enough bandwith is available or will indicate the failure in the RSVP RESV packet.

RSVP has gained a bad reputation of not being scalable. However, Cisco argues, that the bad reputation is because RSVP had been used for Call Admission Control as well as Queuing and Scheduling. When using RSVP for Call Admission Control only it will be scalable and they recommend to use the "DiffServ model" based on packet marking for Queuing and Scheduling.

Classification of Packets

Network elements like routers or switches classify the ingress traffic based on

• ingress interface
• Information found in the frames (Layer 2) like
  • 802.1p/802.1Q priority (often called COS, class of service; mostly used)
  • L2 MAC Addresses
  • VLAN ID
• Information found in the packets (Layer 3) like
  • IP Precedence/DiffServ Codepoints (DSCP) (mostly used)
  • IP Addresses
  • TCP/UDP port numbers
• or higher-layer information (e.g. Cisco NBAR; usually performed at network edges only, and mapped to DiffServ Codepoints there)

The most commonly classification scheme is to use IEEE 802.1p/Q priority bits for layer 2 switching and the Type of Service/DiffServ Codepoints within the IP header for layer 3 queuing.

CoS and DSCP of Voice Streams

For voice (real time protocol, RTP) streams, Siemens phones use the DiffServ Codepoint EF, "Expedited Forwarding" (i.e. decimal value 46, binary 101110) on layer 3. Since most network vendors map the first 3 bits to the IEEE 802.1p/Q priority (CoS) bits, it is recommended to use CoS 5 (binary 101) on layer 2.

A Protocol Implementation Conformance Statement (PICS) for QoS of Siemens products does not exist yet. If in doubt about the used CoS and DSCP values, configure those values manually or consult the product manuals on EDoku.

CoS and DSCP of Signaling Packets

For signaling the DiffServ Codepoint AF31 (decimal value 26, binary 011010) is used. Since most network vendors map the first 3 bits to the IEEE 802.1p/Q priority (CoS) bits, it is recommended to use CoS 3 (binary 011) on layer 2.

A Protocol Implementation Conformance Statement (PICS) for QoS of Siemens products does not exist yet. If in doubt about the used CoS and DSCP values, configure those values manually or consult the product manuals on EDoku.

Class based Packet Processing

If an egress interface is congested, the network element needs to perform an action on packets. It needs to decide, which packets are sent with a higher priority than others, which packets will be stored and queued for later transmissions and which packte will be dropped.

Queuing and Scheduling

In today's most network elements, two ore more packet queues exist per egress interface. Based on the traffic class of a packet, network elements will send the packet to a queue, if the egress interface is congested. The interface's scheduler chooses a packet from a queue and sends it to the egress interface. Several scheduling mechanisms exist:

• priority queuing: as long as the highest priority queue is no empty, packets from this queue are sent only and all other queues need to wait. This can lead to starvation of lower priority packet flows.
• (weighted) round-robin: all queues are served one after the other e.g. queues 1 to 4 are served in the order 1-2-3-4-1-2-3-4 etc. If queue 1 has double weight than the other ones, it will be served like 1-1-2-3-4-1-1-2-3-4- etc. In complex scheduling mechanisms, the packet length will also be taken into account.
• (class-based) fair queuing: packets are classified based on the flow parameters, e.g. source- and destination-IP-addresses and TCP/UDP ports. Idea is to give each flow a fair access to the egress interface. In class-based fair queuing, packets are mapped to traffic classes. Between classes, e.g. weighted round-robin will be performed. Within a traffic class, fair queuing will take place.
• combinations of the above: e.g. Cisco Low Latency Queuing (LLQ) combines priority queuing for real-time applications with class-base weighted fair queuing.

Traffic Shaping

Traffic shaping is used, if transmitting packets at wire speed at one network link would lead to congestion at an other network link. Consider a QoS capable router that is connected via 100 Mbps fast ethernet to a non-QoS-aware DSL modem, which in turn can send with a maximum transmission rate of 1 Mbps. In this case, the router should slow down the transmission rate to 1 Mbps.

The advantage is, that the router can make sure, that high priority traffic will be preferred over other traffic. If the traffic is not shaped within the router, the flow will hit the DSL modem, who will drop packet randomly.

For traffic shaping, the leaky bucket algorithm is used.

Congestion Avoidance

Random Early Detect (RED) is an intelligent packet dropping mechanisms used to control TCP flows in order to avoid TCP global synchronization.

As long as there are no packet drops, TCP tends to increase the flow bandwidth. If the output buffer of a bottleneck link is filled up, without RED, packets of all TCP flows would experience packet drops (i.e. tail drops) at the same time. All TCP flows will slow down quickly, leading to a bad link usage shortly thereafter. Then TCP increases the flow bandwidth again, until the buffer is full again. This cyclic behavior is caused by TCP global synchronization.

RED will randomly drop packets, befor the buffer is full. With that, only few of the TCP flows are slowed down. With that, the bandwith usage of different TCP flows are decoupled from each other and higher sum throughput with less packet drops can be reached.

RED does not work with UDP, since UDP will not slow down transmission rates, when packets are dropped.

Siemens Internal Documentation

• In the EDoku document HiPath 8000 V2.2, Network Planning Guide, Planning Guide, Issue 2 you find chapters about "Network Quality of Service" (chapter 11) and Call Admission Control (chapter 14).
• Oscar document QoS Default Setting Profile V1.0 describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered netinfo users only).

External Links

• QoS on Public Wiki