Alcatel OmniPCX Enterprise

Transcription

1 Alcatel OmniPCX Enterprise QoS for VoIP Overview 1 OBJECTIVE: Describe the essential parameters for QoS

2 The QoS parameters regarding the data network IP Packet Transfer Delay (IPTD): Time for the packet to cross the network IP Packet Delay Variation (IPDV): Variation in the transfer delay of the IP packet IP Packet Loss Ratio (IPLR) : = lost packets / transmitted packets IPDV IPTD 2 IP Device IP network IP Device IPLR 2 IP Packet Transfer Delay = Transit delay IP Packet Delay Variation = Jitter The topic here concerns the carrying of the voice, which is a full duplex type of communication. Every of those parameters will have to be considered in both ways: one way and return.

3 Transit Delay The transit delay has to be considered from end to end Example: Processing delay Play out delay Algorithm delay Packetization delay Serialization delay Propagation delay Component delay Impact on a voice communication Below 100 ms, most users will not notice the delay Between 100 ms and 300 ms, users will notice a slight hesitation in their partner s response Beyond 300 ms, delay is obvious and communications becomes quasi half-duplex communications 3 Codec delay : G ms G ms G ms Packetization delay : Link to the framing period 20 ms usually used. Means that the samples of voice are stored and sent every 20 ms In case of G723.1 the framing is 30 ms Serialization delay : Time necessary to transmit the IP packet. Example with G711 on IP level the payload is 240 Bytes Wan connection time/bit time/packet 64Kb/s 156 micro seconds 30 ms 256Kb/s 3,9 micro seconds 7,5 ms 2Mb/s 0,5 micro seconds 0,96 ms Propagation delay : Time necessary for the signal to travel along the transmission medium (function of the distance). Around 5 microseconds/km in a cable. In case of satellite the delay is around 110 ms for a km altitude satellite and around 260 ms for a km altitude satellite. Component Delay : These are delays caused by the various components within the transmission system. For example a frame passing through a router has to move from the input port to the output port through the backplane. There is some minimum delay due to the speed of the backplane and some variable delays due to queuing and router processing. Processing delay : Time due to the jitter buffer and the play out processing

4 Alcatel devices delay Average of delays introduced by IP boards (GA/GD/INTIP) for one packetisation/depacketisation in G711 :100 ms for one compression/decompression in G723 : 150 ms for one compression/decompression in G729 : 120 ms For transit operation (INTIP only) : 20 ms Average of delays introduced by IP Phones (e-reflexes/x8 serie) for one packetisation/depacketisation in G711 : 60 ms for one compression/decompression in G723 : 90 ms for one compression/decompression in G729 : 70 ms Average of delays introduced in case of communication between an IP phone and a legacy set (VoIP board involved in the communication) for one packetisation/depacketisation in G711 : 80 ms for one compression/decompression in G723 : 120 ms for one compression/decompression in G729 : 100 ms 4 * Transit is not available on GD board nor with G711algorithm.

5 Jitter Variable transit delays means JITTER Reminder: definition of jitter: Packets arriving too early or late compared with the initial rate Burst or gaps in packets. Note: VoIP quality is very sensitive on the timely delivery of packets. => Dejitter buffer to smooth out variable packet reception * 5 * but increases end-to-end delay

6 Jitter (Cont.) Jitter Principle Timestamp Voice sample sende r Net work receiver Network delay Total delay Dejittering delay Packet loss Variable delay Out-of-order arrival time Origin of the jitter network: congestion, queuing delay due the variation in the volume of other traffic streams equipment: capabilities... 6

7 Packet loss Essential parameter to define the quality of voice over IP Loss of 2/3 packets: crackling, metallic voice Sensitivity to packet loss depends on codec (e.g. G711 more sensitive than G723.1 or G729) Origin of packet loss in the network Congestion, queues overflow Router traffic management (packet discarding: RED...) Packet get lost when arriving too late at the receiver Dejitter buffer to avoid packet loss Balance between size of jitter buffer and packet loss to optimize audio quality 7 RED: Random Early Drop. Some routers start to discard packets when the traffic is too high. It was not a problem for data exchange because of the retransmission mechanism but for voice packet it is very bad.

8 Bad Frame Interpolation (BFI) If a packet is lost or corrupted then a BFI (Bad Frame Interpolation) is generated Interpolates lost and/or corrupted packets by using the previously received voice frames for increasing voice quality BFI process is able to estimate how to build the missing packet BFI have impact on communication: 1 BFI: unnoticed 2 consecutive BFI: discrepancies 3 consecutive BFI: silence Remark: if a packet arrives rapidly after a silence, it generates a variation of decibels that will prompt a metallic sound 8 For more information concerning BFI you can check the ITU G113 recommendation about transmissions impairments.

9 Packetization time or Framing The packetization time is the frequency used by the equipment before to send the voice samples The samples are generated according to the frequency used by the algorithm and are stored in a buffer This buffer is empty at the selected framing value If the framing is decreased then the quality is better because the delay is reduced but there is no optimization of the bandwidth and the loading of the equipment is heavy The useful data represent a small part of the frame compared to the header information and there are more frames 9

10 Packetization time or Framing (Cont.) If the framing is increased then the quality is lower because the delay is more important but there is optimization of the bandwidth The useful data represent a more important part of the frame compared to the header information It is possible to increase the framing when the delay is very low in the network Because the framing will add some delay if the network is really reliable packets are bigger and the quantity of voice sample lost is more important That s why the more common values are 20 or 30 ms 10

11 The three quality profiles TOLL-QUALITY = identical to a conventional system Network Round Trip delay < 150ms Jitter < 20 ms Packet loss ratio < 1% NEAR TOLL-QUALITY = some disruptions, inability to recognize tone of voice and crackling (GSM level) Network Round Trip delay < 400ms Jitter < 50 ms Packet loss ratio < 3% BEST-EFFORT QUALITY= metallic sounding voice, loss of syllables and dropouts (marine radio level) Network Round Trip delay < 600ms Jitter < 75 ms Packet loss ratio < 5% 11 Source: AHM Technologies Corporation Algorithmic delay Algorithm G.711 PCM G PCM Delay (ms) Line speed 10 Mbps Ethernet T1 512 Kbs 64 Kbs ASDL ( 640 Kbs) Delay (ms) Propagation delay, e.g ms x distance in km for coax and radio relay systems ms x distance in km for optical fiber systems ms x distance in km for submarine coax systems Component delay, e.g. - vary depending on the component - for example, 1.5 ms for echo cancellers

12 Available bandwidth according to the type of Codec Vocoder Bit rate Packetisa- tion time Packets per second RTP payload size (bytes) IP frame size = payload +RTP(12)+ UDP(8)+IP(20) Bandwith at IP level Bandwith at Ethernet level G Kb/s (MP-MLQ) MLQ) 30 ms Bytes 17.1Kb Kb/s 27.2 Kb/s G.729A 8 Kb/s 30 ms Bytes 18.7 Kb/s 28.8 Kb/s G Kb/s 30 ms Bytes 74.7 Kb/s 84.7 Kb/s FAX 9600 b/s 40 ms Bytes 15.4 Kb/s 23 Kb/s 12 For bandwidth estimation do not take into account the potential gain of VAD. 20% of margin must be added for security. For RTP payload calculation you have to take : -The bit rate of the algorithm that you convert in Bytes : bits /s = 8000 Bytes /s -The framing value : 30 ms -You calculate the frequency per second : 1/0.03 = 33,33 -Then you do the following operation : 8000 / 33,33 = 240 Bytes

13 Echo The echo problem is more often noticed when a communication is made with an analog set or an analog public line Digital sets may introduce echo by acoustic coupling Possible causes for acoustic coupling Hands free feature Speaker feature Digital lines (e.g. ISDN) or VoIP can not cause echo 13

14 Echo (Cont.) IP devices (IP phones, IP gateways) Can not be the cause of the echo (digital components) Compression/decompression (G711, G723.1, G729) may accentuate the echo phenomenon by the introduced delay The embedded echo cancellation feature may not be effective if the echo delay is too long 14

15 Echo (Cont.) Echo cancellation Depth of echo cancellation can not be configured and will be the best the hardware can provide GIP4 4/ 1 and MADA1/MADA3 will use 64 ms GIP6 and MCV8 and MCV 24 boards will use 32 ms 15

16 VoIP quality measurement MOS (Mean Opinion Score) is a subjective method of voice quality measurement There are two test methods: Conversation opinion test Listening-opinion test Test subjects judge the quality of the voice transmission system either by carrying on a conversation or by listening to speech samples. They then rank the voice quality using the following scale: 5 Excellent, 4 Good, 3 Fair, 2 Poor, 1 Bad 16 You can check the ITU-T P.800 recommendation for more details concerning MOS. Coding Standard MOS G (64 kbps) G (8 kbps) G (6.3 kbps) 3.5 (5.3 kbps)

17 VoIP quality measurement (Cont.) MOS is then computed by averaging the scores of the test subjects An average score of 4 and above is considered as tollquality. 17

18 QoS at level 2 and level 3 LAN (level 2) Implement End to End QoS policy Main solution: IEEE 802.1Q (qualification ): traffic separation IEEE 802.1p (priorization): traffic priorization WAN (level 3) Implement End to End QoS policy Main solutions: TOS precedence (TOSp) : traffic priorization DiffServ (DS byte): traffic priorization 18 See network and QoS protocols module for more details

19 QoS at level 2 and level 3 You need to discuss with the LAN administrator to know what type of QoS is implemented in the network To know if you need to provide from the IP devices the QoS information or if the LAN manager has defined the rules directly in the switches and routers 19 See the QoS management module for more information concerning the parameters.

20 20