Media: Voice and Video in your SIP EnvironmentJitendra Shekhawat

Agenda

• Common Audio and Video Codecs • Media/Codec Negotiations • Tuning Your Network for Voice and Video • QoS issues, metrics and user quality

expectations

Objective: Introduction of Media in the SIP environment.

SIP

RTP

SIP

RTP

IP Audio/Video Telephony Network

IPRTP

SIP

SIP Soft Phone

SIP Desk Phone

PC – Email Client

Applications• Video Mail• Video Portal• Live content

streaming

Broadband Users

Multimedia Server

SIP Video Endpoints

SIP Proxy Server

SIP

RTSP Streaming

Server

CNN, ESPN, Bloomberg, live feed

RTP

•Call Control: SIP

•Media: RTP

•Video: H263, H264, MPEG4

•Audio: G711, G723, G729, G726, AMR-NB, etc.

RTSP

SIP Call Example

Audio Video Codecs and Payload Types

• RFC 3551• Some codecs

PT encoding name0 pcmu3 gsm4 g7238 pcma9 g722

18 g72934 h263

Dynamic iLBCDynamic AMRDynamic AMR-WBDynamic AMR-WB+

Media Transport

• RTP – Real Time Transport Protocol– media packet transport– One stream per direction between endpoints

• RTCP– RTP Control Protocol– Provides quality information– Generate reports to the network

RTP Packet

RTP Datagram RTP Datagram RTP Datagram

IP Header 20 bytes

UDP Header 8 bytes

RTP Header 12 bytes

RTP Payload N bytes

Version 2 bits

Padding 1 bit

Extension 1 bit

CSRC count 4

bits

Marker 1 bit

Payload Type 7 bits

Sequence Number 2

bytes

Time stamp 4 bytes

Source Identifier 4 bytes

RTCP Packet

• Receiver of RTP stream sends periodic updates to the originator

• Packet count• Byte count• Packet loss• Timestamps to assess

round-trip delay• Jitter

RTP Packet Payload size

Example: g.711, 20 ms frames: 64000 bps X 20 msec / 8 = 160 byte payload

Payload size =

Function of: codec speed, frame-size

Frequency packets sent

codec speed X frame size

bits/byte

8 X 1000

msec / sec

Media Stream (RTP) Bandwidth:

Packet size := Header + Payload

Header := Ethernet + (IP + UDP + RTP) = 38 + (20 + 8 + 12) = 38 + 40 bytes

Payload := depends on codec

Example: g.711, 20 ms frames (50 packets/s)160 byte payload + (38 + 40) byte header

IP bandwidth: 200 byte/packet = 80,000 bps 160 kbps for 2 way

Ethernet bandwidth: 238 byte/packet = 95,2000 bps 190.4 kbps for 2 way

• Ethernet: Preamble (8) + Ethernet Header (14) + Ethernet CRC (4) + Inter-frame gap (12) = 38

Codec BandwidthsCoder Bitrate Encoded bandwidth

G.711 64 kbps 200-3400 Hz

G.723 5.4 or 6.3 kbps 200-3400 Hz

G.729A (20ms Packet) 8 kbps 200-3400 Hz

AMR 4.75 to 12.2 kbps 200-3400 Hz

AMR-WB Variable: 6.6 up to 23.85 (non-continuous)

50 to 7000 Hz

AMR-WB+ Variable: 6-36 kbps (mono) or 7-48 kbps (stereo)

50 Hz – 7.2 kHz up to 50 Hz – 19.2 kHz

iLBC 13.33 kbps for 30 ms, 15.20 kbps for 20ms

200-3400 Hz

Codec Bandwidths

Coder IP Bandwidth / RTP stream

G.711 (30 ms Packet) 74.6 kbps

G.711 (20ms Packet) 80 kbps

G.711 (10 ms Packet) 96 kbps

G.723.1 (30ms Packet) 15.7 kbps

G.729A (20ms Packet) 24 kbps

AMR (20 ms) 20.4 - 28 kbps

AMR-WB (20ms) 22.4 – 39.6 kbps

AMR-WB+ (20ms) 22 – 52 kbps

iLBC (20ms or 30ms) 31.2 kbps or 24 kbps

Video streams

Frame Sequence

I-frames (Key frames) P-frames (predicted frames)

0

1000

2000

3000

4000

5000

6000

7000

8000

9000

10000

1 18 35 52 69 86 103 120 137 154 171 188 205 222 239 256 273 290 307 324 341 358 375 392 409 426 443 460 477 494 511 528 545 562 579 596 613 630 647 664 681 698

Frame Number

Fram

e S

ize

Video Formats (IP vs. 3G) • High resolution for IP networks

– More bandwidth available– SIP Video Phones are generally CIF size (352 × 288 pixels) – Recommended: CIF, 15 or 30fps, up to 384kbps

• Low resolution for 3G networks– Total bandwidth limited to 64kbps – Generally video + audio is approx 52kbps (12.2kbps AMR + 40kbps

H263) – 3G Mobile phones are generally QCIF size (176 × 144 pixels)

CIF

QCIF

4

3

Performance Issues

• Propagation DelayTime required to travel end to end across the network

• Transport DelayTime required to traverse network equipment

• Packetization DelayTime to digitize, build frames and undo at destination

• Jitter DelayFixed delay by receiver to hold 1 or more packets to damp

variations in arrival times

• Packet LossPacket size impacts sound quality

Jitter Delay

• Calculated on inter-arrival time of successive packets– Average inter-arrival time– Standard deviation

• Goal inter-arrival time = inter-arrival time on emitted packets

• 3 phenomena effecting jitter– Packet loss (threshold 5%)– Silence suppression– Out of sequence packets

• Can be configured on most VoIP equipment

Packet Fragmentation

• Audio RTP packets– Not generally fragmented since packet size is less

than MTU

• Video RTP packets– A large frame is fragmented into a series of packets

for transmission over network– I-Frame fragmentation

• Receiver must receive all fragments to properly reconstruct frame

Packet Loss

• Audio– Packet Loss Concealment (PLC)

• Mask effect of lost or discarded packets• Replay previous packet or use previous packets to estimate missing

data• Key method for improving voice quality

– Packet Loss Recovery (PLR)– Packet Redundancy

• Increased bandwidth

• Video– I-Frame

• If a fragment is lost, subsequent P-Frames will not be sufficient to reconstruct image at receiver

• Video conversion tools available to generate files more suitable for real-time transmission

G.107 to MOS mapping

Codec Bandwidth and Voice Quality Comparison

Codec Payload Bit Rate Voice Quality

G.711 64 Kbps Excellent(MOS 4.2)

G.723 6.4 Kbps / 5.3 Kbps Good (MOS 3.9)

Fair (MOS 3.7)

G.729 8 Kbps Good (MOS 4.0)

G.726 or

G.721

16/24/32/40

Kbps

2/3.2/4/4.2

iLBC 13.33/15.2 kbps Good

(MOS 4.0)

AMR-WB+ 6-36 kbps Good (MOS near 4.0)

Network Issues?

Network Issues – Now What

• Determine the source of delay– Codec’s?– Too many hops?– Not enough bandwidth?

• Define means to reduce delay– Provision smaller packet sizes– Reduce hop count– Change routing protocols used

• Keep monitoring– Find problems first– Objectively identify issues

IP Header

Traffic Shaping

• DiffServ• RSVP• MPLS

Conclusion

• Reliability– Can calls be made when needed? – Will call setup time match current environment?– Will calls be disconnected?

• Quality– Is the voice quality of the calls the same?– Can the users tell the difference?

• Cost– What are the cost benefits of VoIP?– What equipment will be needed?

Wrap-up

Q & A / Quiz

Frame Sizes

Format Dimension (H x W, pixels)

>1 bits/pixel

Sub-QCIF (SQCIF) 128 x 96

Quarter-CIF (QCIF) 176 x 144

CIF (Common Intermediate Format)

352 x 288

4CIF (4 x CIF) 704 x 576

16CIF (16 x CIF) 1408 x 1152