Media: Voice and Video in your SIP Environment Jitendra 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 IP RTP SIP SIP Soft Phone SIP Desk Phone PC – 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

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 IP Header 20 bytes UDP Header 8 bytes RTP Header 12 bytes RTP Payload N bytes Version 2 bits Padding 1 bit Extensio n 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: 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 + ( ) = bytes Payload := depends on codec Example: g.711, 20 ms frames (50 packets/s) 160 byte payload + ( ) byte header IP bandwidth: 200 byte/packet = 80,000 bps  160 kbps for 2 way Ethernet bandwidth: 238 byte/packet = 95,2000 bps  kbps for 2 way Ethernet: Preamble (8) + Ethernet Header (14) + Ethernet CRC (4) + Inter-frame gap (12) = 38

Codec Bandwidths CoderBitrateEncoded bandwidth G kbps Hz G or 6.3 kbps Hz G.729A (20ms Packet)8 kbps Hz AMR4.75 to 12.2 kbps Hz AMR-WBVariable: 6.6 up to (non- continuous) 50 to 7000 Hz AMR-WB+Variable: 6-36 kbps (mono) or kbps (stereo) 50 Hz – 7.2 kHz up to 50 Hz – 19.2 kHz iLBC13.33 kbps for 30 ms, kbps for 20ms Hz

Codec Bandwidths CoderIP 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 (30ms Packet)15.7 kbps G.729A (20ms Packet)24 kbps AMR (20 ms) 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)

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 Delay Time required to travel end to end across the network Transport Delay Time required to traverse network equipment Packetization Delay Time to digitize, build frames and undo at destination Jitter Delay Fixed delay by receiver to hold 1 or more packets to damp variations in arrival times Packet Loss Packet 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 CodecPayload Bit RateVoice Quality G KbpsExcellent (MOS 4.2) G Kbps / 5.3 KbpsGood (MOS 3.9) Fair (MOS 3.7) G.7298 KbpsGood (MOS 4.0) G.726 or G /24/32/40 Kbps 2/3.2/4/4.2 iLBC13.33/15.2 kbpsGood (MOS 4.0) AMR-WB+6-36 kbpsGood (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 FormatDimension (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 CIF (16 x CIF)1408 x 1152