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VoIP

Pulse Code Modulation (PCM)

Marcelo Zanata
Best Codec Use
G.729 WAN with QoS iLBC Network without QoS

Overhead
Ethernet PPP Frame-Relay MLPPP MPLS WLAN IP UDP RTP 18 bytes 6 Bytes 6 Bytes 10 Bytes 4 Bytes 24 Bytes 20 bytes 8 Bytes 12 Bytes

Sampling: 8000 discrete signal measurements are taken at equal intervals every second Quantization: The level of each sample is rounded to the nearest expressible value Encoding: Digital values are encoded as binary numbers for encapsulation Compression (Optional): The signal is compressed in realtime to consume less bandwidth

Layer 2 Qos Marking


MOS G.722 SB-ADPCM4.13 G.711 PCM4.1 iLBC4.1 G.729 CS-ACELP3.92 G.726 ADPCM3.85 G.729a CS-ACELP3.7 G.728 LD-CELP3.61 Bandwidth 48-64 kbps 64 15.2 8 32 8 16 Complexity Medium Low High High Medium Medium High Free Yes Yes Yes No Yes No No

G.729
8 kbps conjugate-structure algebraic code-excited linear prediction (CS-ACELP). G.729a is a compatible extension of G.729, but requires less computational power. G.729 has been extended in Annex B (G.729b) which provides a silence compression method that enables a voice activity detection (VAD) module

Bandwidth Calc
voice payload in bytes = [(codec bit rate in kbps) * (sampling rate in msec)] / 8 packets per second = [1 / (sampling rate in msec)] * 1000 L3 Bandwidth = [(packets per second) * (voice payload + 40 bytes for RTP/UDP/IP headers) * 8 bits] / 1000 L2 Bandwidth = [(packets per second) * (voice payload + 40 bytes for RTP/UDP/IP headers + Y bytes for Layer 2 overhead) * 8 bits] / 1000

G.711
64 kbps g711alaw - for E1 g711ulaw - for T1

Bandwidth Consumption for Voice Payload and IP Header Only


Codec G.711 and G.722-64k G.711 and G.722-64k (SRTP) G.711 and G.722-64k G.711 and G.722-64k (SRTP) iLBC iLBC (SRTP) iLBC iLBC (SRTP) G.729A G.729A (SRTP) G.729A G.729A (SRTP) Medium Complexity G.711 (a-law, mu-law) Fax/modem passthrough Clear channel G.726 (32K, 24K, 16K) Fax relay G.729 (a, ab) Sampling (ms) Voice Payload Pkts/Sec 20 20 30 30 20 20 30 30 20 20 30 30 160 164 240 244 38 42 50 54 20 24 30 34 50 50 33.3 33.3 50 50 33.3 33.3 50 50 33.3 33.3 Bandwidth per conversation 80 kbps 81.6 kbps 74.7 kbps 75.8 kbps 31.2 kbps 32.8 kbps 24.0 kbps 25.1 kbps 24.0 kbps 25.6 kbps 18.7 kbps 19.8 kbps

RTP Payload Type


VAD = 13 RTP-NTE = 101 G.711alaw = 8

Codecs Supported by c5510 DSP


High Complexity G.711 (a-law, mu-law) Fax/modem passthrough Clear channel G.726 (32K, 24K, 16K) Fax relay G.729 G.729 (a, b, ab) G.728 G.723.1 (32K, 24K, 16K) G.723.1a (5.3K, 6.3K) Modem relay Flex Mode At 15 MIPS per call: G.711 (a-law, mu-law) Fax/modem passthrough Clear channel At 30 MIPS per call: G.726 (32K, 24K, 16K) Fax relay G.729 G.729 (a, b, ab) At 40 MIPS per call: G.728 G.723.1 (32K, 24K, 16K) G.723.1a (5.3K, 6.3K) Modem relay

VoIP
Codecs Supported by c5510 DSP
Low Complexity Medium Complexity 3 6 12 18 24 MIPS per PVDM 120 240 480 720 960

Marcelo Zanata
IP Phone Boot Process

PVDM2-8 (1/2 DSP) PVDM2-16 (1 DSP) PVDM2-32 (2 DSP) PVDM2-48(3 DSP) PVDM2-64(4 DSP)

4 8 16 24 32

1. Power Over Ethernet (Optional) Power is supplied via IEEE 802.3af/at or Cisco Low Complexity Medium High Very High ILP Complexity Complexity Complexity 2. VLANs Learned via CDP or LLDP G.711 (a-law, mu-law) G.726 G.729 iSAC Voice and data VLANs communicated via Fax Passthrough Fax Relay G.729B CDP/LLDP Modem Passthrough G.729A G.723 3. IP Assignment via DHCP Clear channel G.729AB G.728 The phone sends a DHCP request in the voice G.722 Modem Relay VLAN; the response includes an IP and DHCP GSMFR iLBC option 150 GSMEFR 4. Configuration Retrieved via TFTP Codecs Supported by PVDM3 The phone retrieves its configuration from Low Medium High Very High one of the TFTP servers specified in the DHCP Complexity Complexity Complexity Complexity option 5. Registration PVDM3-16 16 12 10 8 The phone registers with the call server(s) PVDM3-32 32 22 14 12 specified PVDM3-64 64 44 28 24 in its configuration PVDM3-128 128 97 60 50 IP Phone auto-registration PVDM3-192 193 140 89 74 PVDM3-256 258 194 121 101 The same stetp 1,2,3 of boot process CDP 1. Phone try to get via TFTP [mac].cnf.xml configuration file. If the file does not exists, Multicast 01-00-0C-CC-CC-CC they go to step 2, also go to registration Sent every 60 secs 2. Phone try to get XMLDefault.cnf.xml. Work on Ethernet, Frame Relay, ATM CUCM assign a directory number and create Holdtime of 180 secs (to delete from neighbors list) the records to this phone, then create Carrier the informations: Operating Syste / Hostname / IP Address / Port ID / Device Type,Model / Duplex Settings / VTP Domain / Native VLAN / Power Draw [mac].cnf.xml 3. Phone updates their firmware if necessary Encoded as TLV (Type-Length-Value) AutoQoS uses CDP Switch send QoS Values (CoS) to used on PC Port of ephone 4. Phone register on server specified on XMLDefault.cnf.xml TAPS Only one CCM Group will have the The same step of auto-reg auto-reg enabled 1. User dial to CTI Route Point assigned to CRS application Voice Class Configuration 2. User enter the appropriate Directory Number voice class codec 1 3. The number is looked up in the phone configuration records that were codec preference 1 g729r8 previously added using a dummy MAC address. codec preference 2 g711ulaw ! 4. Cisco CRS update the dummy MAC with the actual MAC dial-peer voice 9 5. Phone reset and download its newly configuration from TFTP Server voice-class codec 1

Codecs Supported by PVDM3

Allow inter-VoIP Connections


voice-service voip allow-connections [h323/sip] to [h323/sip]

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