QoS - Revision history

Christian.kaehler: /* brand update, unresolved references*/

2015-04-29T07:19:01Z

‎brand update, unresolved references

Horemu at 12:38, 25 April 2011

2011-04-25T12:38:57Z

Hans.zeiner at 14:32, 19 October 2009

2009-10-19T14:32:56Z

Tm24361 at 15:26, 31 October 2008

2008-10-31T15:26:37Z

Tm24361 at 15:25, 31 October 2008

2008-10-31T15:25:04Z

Tm24361 at 15:21, 31 October 2008

2008-10-31T15:21:05Z

Horemu: /* Classification of Packets */

2007-09-27T07:49:08Z

‎Classification of Packets

Horemu: /* Classification of Packets */

2007-09-27T07:48:57Z

‎Classification of Packets

Horemu: /* Classification of Packets */

2007-09-27T07:48:25Z

‎Classification of Packets

Horemu at 07:49, 5 September 2007

2007-09-05T07:49:43Z

@@ Line 25: / Line 25: @@
 === Centralized Static Call Admission Control ===
-Siemens [[OpenScape Voice]] 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.
+Unify [[OpenScape Voice]] 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 [[OpenScape Voice]] solution, telephones are classified to be members of certain zones based on IP-addresses or telephone numbers.
@@ Line 57: / Line 57: @@
 === 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.
+For voice (real time protocol, RTP) streams, Unify 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.
@@ Line 97: / Line 97: @@
 * [http://en.wikipedia.org/wiki/Quality_of_service QoS on Public Wiki]

← Older revision		Revision as of 12:38, 25 April 2011
Line 1:		Line 1:
−	~~<div class="breadcrumb">~~	+	{{Breadcrumb\|0\|0\|Glossary}}
−	{{up}} ~~[[Glossary]]~~
−	~~</div>~~

	'''QoS''' = '''Q'''uality '''o'''f '''S'''ervice		'''QoS''' = '''Q'''uality '''o'''f '''S'''ervice

@@ Line 27: / Line 27: @@
 === 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.
+Siemens [[OpenScape Voice]] 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.
+In the case of the [[OpenScape Voice]] 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.
@@ Line 102: / Line 102: @@
 == Siemens Internal Documentation (non-public) ==
-* In the [https://netinfo2.icn.siemens.de/edoku3/search_de.htm EDoku] document  [http://netinfo2db.icn.siemens.de:8080/techdoc/en/P31003H8022P1020176A9/wwhelp/wwhimpl/js/html/wwhelp.htm?href=wwh3master.html 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).
+* In the [https://netinfo2.icn.siemens.de/edoku3/search_de.htm EDoku] document  [http://netinfo2db.icn.siemens.de:8080/techdoc/en/P31003H8022P1020176A9/wwhelp/wwhimpl/js/html/wwhelp.htm?href=wwh3master.html OpenScape Voice 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).
 * [https://intranet.enterprise.siemens.com/Content%20Sites%20V2/Corporate/SEN_The_Portfolio/Research_and_Development/Enterprise_Systems_CTO/Information_Pool/Interfaces_and_Protocols.aspx?tab=1&sc_lang=en Oscar] document [https://www.g-dms.com/livelink/livelink.exe?func=ll&objId=21539133&objAction=download&viewType=1 QoS Default Setting Profile V1.0] describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered G-DMS users only).
 * [[Cisco QoS Cookbook for LAN]]

@@ Line 103: / Line 103: @@
 * In the [https://netinfo2.icn.siemens.de/edoku3/search_de.htm EDoku] document  [http://netinfo2db.icn.siemens.de:8080/techdoc/en/P31003H8022P1020176A9/wwhelp/wwhimpl/js/html/wwhelp.htm?href=wwh3master.html 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).
-* [https://intranet.enterprise.siemens.com/Content%20Sites%20V2/Corporate/SEN_The_Portfolio/Research_and_Development/Enterprise_Systems_CTO/Information_Pool/Interfaces_and_Protocols.aspx?tab=1&sc_lang=en Oscar] document [https://www.g-dms.com/livelink/livelink.exe?func=ll&objId=21539133&objAction=download&viewType=1 QoS Default Setting Profile V1.0] describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered netinfo users only).
+* [https://intranet.enterprise.siemens.com/Content%20Sites%20V2/Corporate/SEN_The_Portfolio/Research_and_Development/Enterprise_Systems_CTO/Information_Pool/Interfaces_and_Protocols.aspx?tab=1&sc_lang=en Oscar] document [https://www.g-dms.com/livelink/livelink.exe?func=ll&objId=21539133&objAction=download&viewType=1 QoS Default Setting Profile V1.0] describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered G-DMS users only).
 * [[Cisco QoS Cookbook for LAN]]

@@ Line 103: / Line 103: @@
 * In the [https://netinfo2.icn.siemens.de/edoku3/search_de.htm EDoku] document  [http://netinfo2db.icn.siemens.de:8080/techdoc/en/P31003H8022P1020176A9/wwhelp/wwhimpl/js/html/wwhelp.htm?href=wwh3master.html 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).
-* [https://intranet.enterprise.siemens.com/Content%20Sites%20V2/Corporate/SEN_The_Portfolio/Research_and_Development/Enterprise_Systems_CTO/Information_Pool/Interfaces_and_Protocols.aspx?tab=1&sc_lang=en Oscar] document [https://netinfo3.icn.siemens.de/projects/oscar/specifications/oscar40/Chap03Scenarios/348QoSValues_v1_0.doc QoS Default Setting Profile V1.0] describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered netinfo users only).
+* [https://intranet.enterprise.siemens.com/Content%20Sites%20V2/Corporate/SEN_The_Portfolio/Research_and_Development/Enterprise_Systems_CTO/Information_Pool/Interfaces_and_Protocols.aspx?tab=1&sc_lang=en Oscar] document [https://www.g-dms.com/livelink/livelink.exe?func=ll&objId=21539133&objAction=download&viewType=1 QoS Default Setting Profile V1.0] describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered netinfo users only).
 * [[Cisco QoS Cookbook for LAN]]

@@ Line 55: / Line 55: @@
 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.
-[[Image:QoS_Frame_Packet_500x265.jpg|400px|Highlighting CoS and DSCP in an Ethernet Frame (red) with an embetted IP Packet (blue)]]
+[[Image:QoS_Frame_Packet_500x265.jpg|Highlighting CoS and DSCP in an Ethernet Frame (red) with an embetted IP Packet (blue)]]
 === CoS and DSCP of Voice Streams ===

@@ Line 45: / Line 45: @@
 * Information found in the frames (Layer 2) like
 ** 802.1p/802.1Q priority (often called COS, class of service; mostly used)
-** L2 MAC Addresses
+** L2 [[MAC]] Addresses
 ** VLAN ID
 * Information found in the packets (Layer 3) like

@@ Line 1: / Line 1: @@
-== General ==
+<div class="breadcrumb">
-'''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
+'''QoS''' = '''Q'''uality '''o'''f '''S'''ervice
 * Call Admission Control,
 * Classification of Packets,
@@ Line 16: / Line 20: @@
 == 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.
@@ Line 21: / Line 26: @@
 === 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.
@@ Line 28: / Line 34: @@
 === Network based dynamic Call Admission Control ===
 A standard way of performing Call Admission Control is [http://en.wikipedia.org/wiki/Resource_Reservation_Protocol Ressource reSerVation Protocol (RSVP)], i.e. the [http://en.wikipedia.org/wiki/Intserv 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.
@@ Line 33: / Line 40: @@
 == Classification of Packets ==
 Network elements like routers or switches classify the ingress traffic based on
 * ingress interface
@@ Line 50: / Line 58: @@
 === 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.
 === 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.
 == 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.
@@ Line 75: / Line 87: @@
 === Congestion Avoidance ===
 '''Random Early Detect (RED)''' is an intelligent packet dropping mechanisms used to control TCP flows in order to avoid [http://en.wikipedia.org/wiki/TCP_global_synchronization TCP global synchronization].
@@ Line 83: / Line 96: @@
 RED does not work with UDP, since UDP will not slow down transmission rates, when packets are dropped.
-== Siemens Internal Documentation ==
+== External Links ==
 * In the [https://netinfo2.icn.siemens.de/edoku3/search_de.htm EDoku] document  [http://netinfo2db.icn.siemens.de:8080/techdoc/en/P31003H8022P1020176A9/wwhelp/wwhimpl/js/html/wwhelp.htm?href=wwh3master.html 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).
 * [http://telecom.gud.siemens.at/oscar/ Oscar] document [https://netinfo3.icn.siemens.de/projects/oscar/specifications/oscar40/Chap03Scenarios/348QoSValues_v1_0.doc QoS Default Setting Profile V1.0] describes default CoS (802.1p) and DSCP values in Siemens phones/clients and gateways (registered netinfo users only).
 * [[Cisco QoS Cookbook for LAN]]