These are partly notes I have jotted down from what I am reading and also links to helpful VOIP sites. I lay no claims to ownership, except where indicated.

Sunday, September 26, 2010

Step-By-Step Linux install guide

Step-By-Step Install Guide Linux CentOS-5 Server -

Thursday, February 4, 2010

Understanding Call-Types

Call Types:


does not traverse the WAN or PSTN


Occurs between two tels on same data ntwk



Occurs when a user dials an access code (9) to connect to the PSTN

FXO. E1 or T1.


Auto connects one tel to another when the first tel goes off-hook

can work btw any type of signaling inc, E&M, FXO, FXS or any combo of ana and digi interfaces.


Originates at one PBX and terminates at another

form of toll-bypass

-Intercluster trunk calls

occurs when calls are routed by two separate CUCMs


when calls originates on an internal ntwk and are routed to an external ntwk (PSTN)


Local call- One staff member calls another staff member at the same office. The
call is switched between two ports on the same voice-enabled router.

On-net call -One staff member calls another staff member at a remote office. The
call is sent from the local voice-enabled router, across the IP network,
and is terminated on the remote office voice-enabled router.

Off-net call - A staff member calls a client who is located in the same city. The call is sent from the local voice-enabled router, which acts as a gateway, to
the PSTN. The call is then sent to the PSTN for call termination.

PLAR call - A client picks up a customer service telephone located in the lobby of
an office and is automatically connected to a customer service representative
without dialing any digits. The call is automatically dialed based on the PLAR configuration of the voice port. In this case, as soon as the handset goes off hook, the voice-enabled router generates the prespecified digits to place the call.

PBX-to-PBX call - One staff member calls another staff member at a remote office. The
call is sent from the local PBX, through a voice-enabled router, across
the IP network, through the remote voice-enabled router, and terminated
on the remote office PBX.

Intercluster trunk call - One staff member calls another staff member at a remote office using IP phones. The call setup is handled by a Cisco Unified
Communications Manager server at each location. After the call is set
up, the IP phones generate IP packets carrying voice between sites.

On-net to off-net call - One staff member calls another staff member at a remote office while the IP network is congested. When the originating voice-enabled
router determines that it cannot complete the call across the IP network,
it sends the call to the PSTN with the appropriate dialed digits
to terminate the call at the remote office via the PSTN network.

Three types of Analog Voice Interfaces supported by Cisco devices:


connects directly to end-user equip - tels, fx, mods


used for trunk or tie line connection to a PSTN or PBX that does not support E&M signaling.


preferable to above options as it delivers better answer and disconnect supervision.

E-lead and M-lead


1. Supervisory

detection of changes to the status of a loop or trunk


no way to prevent GLARE - CO and subscriber seize same line, same time - use ground-start to prevent glare.

does not provide switch-side disconnect supervision for FXO calls.


2. Addressing

passing dialed digits - tone or pulse - to PBX or CO. These digits provide path to CPE

-pulse dialing

-dtmf dialing

3. Informational

audible tones to use (incoming calls etc)

-call progress tones


mainly used btw PBXs or network to network telephony switches.

Types I, II, III, V supported

SSDC5 mainly found in England


3 E&M access types:

-Immediate-Start signaling

simplest method
suffers from glare

The calling side seizes the line by going off-hook
on its E lead, waits for a minimum of 150 ms and then sends address information as
DTMF digits or as dialed pulses. This signaling approach is used for E&M tie trunk

-wink-start signaling

most commonly used

In wink-start, the calling
side seizes the line by going off-hook on its E lead; it then waits for a short temporary off-hook pulse, or “wink,” from the other end on its M lead before sending
address information as DTMF digits. The switch interprets the pulse as an indication
to proceed and then sends the dialed digits as DTMF or dialed pulses. This signaling
is used for E&M tie trunk interfaces. This is the default setting for E&M voice ports.

-delay-start signaling:

the calling station seizes the line by going off-hook on its E lead. After a timed interval, the calling side looks at the status of the called side. If the called side is on-hook, the calling side starts sending information as DTMF digits. Otherwise, the calling side waits until thecalled side goes on-hook and then starts sending address information. This signaling approach is used for E&M tie trunk interfaces.






Are used to interconnect GWs or PBX to other GWs.

can be analog or digital

Signaling can be done using either the voice channel (in-band) or an extra dedicated channel (outof-band). The available features depend on the signaling protocol in use between the devices.

Centralized Automated Message Accounting (CAMA)

Five options exist:
■ KP-0-NXX-XXXX-ST: 7-digit ANI transmission. The Numbering Plan Area (NPA),
or area code, is implied by the trunk group and is not transmitted.
■ KP-0-NPA-NXX-XXXX-ST: 10-digit transmission. The E.164 number is fully
■ KP-0-NPA-NXX-XXXX-ST-KP-YYY-YYY-YYYY-ST: Supports CAMA signaling with
ANI/Pseudo ANI (PANI).
■ KP-2-ST: Default transmission when the CAMA trunk cannot get a corresponding
Numbering Plan Digit (NPD) in the look-up table or when the calling number is
fewer than 10 digits. (NPA digits are not available.)
■ KP-NPD-NXX-XXXX-ST: 8-digit ANI transmission, where the NPD is a single MF
digit that is expanded into the NPA. The NPD table is preprogrammed in the sending
and receiving equipment (on each end of the MF trunk). For example: 0=415, 1=510,
2=650, 3=916
05551234 = (415) 555-1234, 15551234 = (510) 555-1234
The NPD value range is 0–3.
When you use the NPD format, the area code needs to be associated with a single digit.
You can preprogram the NPA into a single MF digit using the ani mapping voice port
command. The number of NPDs programmed is determined by local policy as well as by
the number of NPAs the PSAP serves. Repeat this command until all NPDs are configured
or until the NPD maximum range is reached.

Sunday, January 31, 2010

Jots for today

Understanding Codecs and DSP functionality



-a device or program capable of performing encoding or decoding on some digital data stream or signal.

-transform voip media streams into another format: A to D; D to D; or D to A.

-especially important on low-speed serial links where bandwidth is very important.

Codecs supported by the Cisco IOS GWs:


for encoding tel audio on 64-kbps channel

it is a PCM scheme operating at 8KHz sample rate, with 8 bits per sample

widely used in telecoms ind as it improves the signal-to-noise ratio without increasing the amount of data.

has two subsets:

1. mu-law

used in Nth Ame and Jap phone ntwks.

2. a-law

used in Europe and elsewhere around the world.

both subsets use compressed speech carried in 8-bit samples. Use 8KHz sampling rate with 64kbps of storage.


wideband speech codec.
provides 7KHz of wideband audio at data rates from 48kbps to 64kbps.

tech is based on adaptive differential PCM (ADPCM).

G.722.1 - lower bit-rate compressions

G.722.2 (Adaptive Multi-Rate Wideband)- offers even more lower bit-rate compressions


is an ITU-T ADPCM which ops at data rates of 40, 32, 24, 16 kbps.


16kbps low-delay CELP (LD-CELP)


uses conjugate-structure algebraic-CELP (CS-ACELP)


describes dual-rate coder for multimedia communications for compressing speech or audio signal components at very low-bit rate as part of the H.324 family of standards.

two bit rates assoc with it:

r63: 8.3 kbps using 24 byte frames and Multipulse LPC with Maximum Likelihood Quantization (MPC-MLQ)

r53: 5.3 kbps using 20 byte frame and the ACELP algorithm.


frame size of 20 ms and ops at 13 kbps bit rate.

is a Regular Pulse Excited-Linear Predictive (RPE-LTP) coder.

network must support GSM FR codecs in order to write VoiceXML scripts.

-iLBC [Internet Low Bit Rate Codec]

designed for narrowband speech
results in a payload bit rate of 13.33kbps for 30 ms frame and 15.20 kbps for 20 ms frames.

algorithm is based on block-independent linear predictive coding with the choice of data frame lengths of 20 ms and 30 ms.

There is a need to balance the need for voice quality against the cost of BW when choosing codecs. The higher the codec BW, the higher the cost of each call across the ntwk.


-voice sample size is a variable that can affect total BW used.

-voice sample is defined as the total output from a codec DSP that is encapsulated into a Protocol Data Unit (PDU)

Table of various codecs, their sample sizes and the number of pkts reqd for voip to xmit 1 second of audio:

The larger the sample size, the larger the packet and the fewer the encapsulated samples that have to be sent (wc reduces BW).



Bytes_per_Sample = (Sample_Size * Codec_Bandwidth) /8

Other factors to bear in mind when calculating overhead of voip call:

1. Layer 2 and security protocols add to pkt size significantly.

Layer 2 overhead for various protocols:

-Ethernet II protocol

carries 18 bytes of overhead
6 for source MAC
6 for dest MAC
2 for type
4 for CRC


carries 6 bytes of overhead

1 flag byte to indicate beginning and end of a frame

1 address byte

1 control byte

1 protocol byte

2 bytes for CRC


carries 6 bytes of overhead

2 bytes for DLCI header

2 for FRF.12

2 for CRC

-Multilink PPP

carries 6 bytes of overhead

1 for flag

1 for address

2 for control or type

2 for CRC

2. The IP and transport layers also contribute to overhead

IP adds a 20 byte header

UDP adds 8 byte header

RTP adds a 12 byte header

3. Security overhead

IPSec adds 50 to 57 bytes of overhead when u r using VPN.

L2TP or GRE adds 24 bytes of overhead.

if in use, MLP will add 6 bytes

MPLS adds 4 byte label to every pkt


Points to consider before calculating:

-if more bw is reqd for the codec, then more total bw is reqd.

-if more overhead is assoc with the data link, the more total bw is needed.

-if there is a larger sample size, then less total bw is reqd.

-if cRTP is being used then the total bw reqd is reduced significantly.

As a ntwk engineer:

-you need to calc the total bw for each voip call

-this info can then be used to calculate the total bw for the company's WAN links


TBW = total packet size * packets per second (pps)

-total packet size in bytes = (Layer 2 header: MPPP, FRF.12, or Ethernet) + (IP/UDP/RTP header)+(voice payload size)

-pps = codec bit rate/voice payload size

Protocol header assumptions used for the calcs:

-40 bytes for IP(20)/UDP(8)/RTP(12) headers

-cRTP reduces IP/UDP/RTP to 2 or 4 bytes (cRTP not available over Ethernet)

-6 bytes for MPPP, or FRF.12 L2 header

-1 byte for the end-of-frame flag on MP or Frame Relay frames

-18 bytes for Ethernet L2 headers (including 4 bytes for FCS or CRC)

Example calc:

G.729 codec (8 kbps) with a 20 byte sample size and using FRF.12 without cRTP

total packet size = 6 bytes (FRF.12) + 40 bytes (IP/UDP/RTP) + 20 bytes (voice payload size) = 66 bytes

total packet size (bits) = 66 bytes * 8 bits per byte = 528 bits

PPS = 8 kbps/160 bits (20 bytes * 8 bits) = 50 p/s i.e 8000/160

BW per call = 528 bits/s * 50 p/s = 26, 400 bps = 26.4kbps.

Example calc:

G.729 codec (8 kbps) with a 20 byte sample size and using FRF.12 with cRTP

total packet size = 6 bytes + 2 bytes + 20 bytes = 28 bytes * 8 bits = 224 bits

pps = 8 kbps/160 bits (20 bytes * 8 bits)= 50 p/s

total BW per call = 224 bits/s * 50 p/s = 11200 bps = 11.2 kbps

VAD provides a max 35% BW savings. VAD should however not be taken into account for the purpose of ntwrk design and bw engineering. Features such as music on hold (MOH) and fax render VAD ineffective. VAD reduces silence on voip conversations but also provides comfort noise generation (CNG).


media resource - sw-based or hw-based entity that performs media processing functions on the data streams to wc it is connected.

-transcoding: conversion from one codec to another.

processed by DSPs on a DSP farm - sessions are initiated and managed by CUCM which refers to transcoders as hw MTPs.

-voice termination: the digitization and packetization of an analog signal on a TDM interface.

-conference bridge: a resource that joins multiple parties into a single call.

hardware conference bridges are used in two environs:

central site

remote site

-MTP: an entity that accepts two full-duplex voice streams using the same codec.

can be used for:

repacketization - transcode a-law to mu-law and vice versa

H.323 supplementary services

two types of MTPs:

1. sw MTP

2. hw MTP


This refers to the amount of processing required to perform voice compression

Two categories:

medium complexity

high complexity

Wednesday, January 27, 2010

Jots for today





This is the degree to wc a sys or a portion of a sys accurately reproduces, at its output, the essential characteristics of the signal impressed upon its input or the result of a prescribed operation on the signal impressed upon its input.


This is the result of electrical impedance mismatches on the transmission path.
Echo is always present.

Two components affect echo:
-amplitude - loudness of the echo
-delay - time between spoken voice and echoed sound

Can be controlled using suppressors or cancelers.


this is the variation in the arrival of coded speech packets at the far end of the VOIP network.

varying arrival times can cause gaps in re-creation/playback of voice-signal and are undesirable and annoy the listener.

network congestion, improper queuing or config errors can cause the unevenness in a steady stream of sent pkts.

use playout delay buffer or dejitter buffer to resolve this by buffering the pkts and playing them out in a steady stream to the digital signal proc (DSP) to be converted back to an analog audio stream. This buffer however affects overall absolute delay which can affect VOIP. Entire words in a conversation can be cut off.


This is the time between spoken voice and the arrival of electronically delivered voice at the other end.

Results from multiple factors including distance (propagation delay), coding, compression, serialization and buffers.

Two types of delay:

-fixed delays

Fixed delay components are predictable and add directly to the overall delay on the connection. These components include:

-coding - the time it takes to translate an audio signal into a digital signal.

-packetization - the time it takes to put digital voice info into pkts and remove the info from the pkts.

-serialization - the insertion of bits into a link

-propagation - the time it takes a pkt to traverse a link.

-variable delays

These arise from queuing delays in the egress trunk buffers that are located on the serial port connected to the WAN.

3 bands of one-way delay are specified by the ITU-T recomm G.114, ie, the acceptable network delay for voice apps:

1. 0 - 150 ms

acceptable for most user apps

2. 150 - 400 ms

acceptable as long as admins are aware of the transmission time and its impact on the transmission quality of user apps.

3. 400 ms

Above 400 ms is unacceptable for general network planning purposes.
might be exceeded in some exceptional circumstances.

G.114 recomm is for conns that are adequately controlled, implying that echo cancelers are in use. Echo cancelers are reqd when one-way delay exceeds 25ms.

For private networks a 200ms goal is reasonable and 250ms delay is the limit.

-Packet loss

Unstable network, network congestion or too much variable delay in the network, may cause voice pkts to be dropped. Lost voice pkts are not recoverable since retransmission is not an option thus leading to gaps in conversations.

Industry standard codec algorithms used in Cisco DSPs will correct for 20 to 50 ms of lost voice packets thru the use of Packet Loss Concealment(PLC) algorithm which analyzes missing packets and generates a reasonable replacement packet to improve the voice quality. Cisco VOIP tech uses 20 ms samples of voice payload per VOIP pkt as default.

Only a single packet can be lost at any time, if more, then gaps are experienced.
-Side tone

This refers to the purposeful design of the tel that allows the user to hear the spoken audio from the ear-piece. Without this, it would seem as if the tel was faulty.

-Background noise

This is the low-volume audio that is heard from the far-end of the connection.

Bandwidth saving devices such as voice activity detection (VAD) can eliminate background noise altogether.


Audio quality can be measured using:

1. MOS:

Mean opinion score (MOS) defined in ITU-T recomm P.800 - average opinion of group of testers is calculated to create MOS score.

Uses subjective testing




2. PSQM:

Perceptual Speech Quality Measurement (PSQM).Automated method for measuring speech quality ''in service'' or as it goes along. Defined in ITU P.861. Usually resides with IP call management systems sometimes integrated into SNMP systems.

Has over 90% accuracy.

Both of these measurement methods are not recommended for modern voip networks. they do not measure problems such as jitter and delay.

3. Perceptual Evaluation of Speech Quality (PESQ):

Cisco Qos Mechs:

use to provide effective priority service.

provided thru IP traffic management, queuing, and shaping.

Cisco IOS features that address reqs of e2e QoS and service differentiation of voice pkt delivery:

-Header compression: - results in decreased consumption of available bandwidth, reduction in delay is also realized. (RTP or TCP header compression)

-Frame relay traffic shaping (FRTS): delays excess traffic using a buffer or queuing mech, to hold pkts and shape the flow when the data rate of the source is higher than expected.

-FRF.12 or higher: ensures predictability for voice traffic, aiming to provide better thruput on low-speed Frame Relay links by interleaving delay-sensitive traffic on one virtual circuit (VC) with fragments of a long frame on another VC utilizing the same interface.

-PSTN fallback: provides a mech to monitor congestion in the IP network and either redirect calls to the PSTN or reject calls based on the network congestion.

-IP RTP Priority and Frame Relay IP RTP Priority: provide a strict priority queuing scheme that allows delay-sensitive data such as voice to be dequeued and sent before pkts in other queues are dequeued.

Works with WFQ and CBWFQ.

Useful of slow WAN links like FR, Multilink PPP (MLP), T1 ATM.

-IP-to-ATM class of service (CoS):

-Low latency queuing (LLQ):

-Multilink PPP (MLP):

-Resource Reservation Protocol (RSVP):

QoS features provide improved and more predictable network operations by implementing these features:

-support guaranteed bandwidth - QoS designs the ntwk in such a way that necessary bandwidth is always available to support voice and data traffic.

-improve loss characteristics - design FR network such that discard eligibility is not a factor when it comes to frame pkts, keeping voice below the CIR.

-avoid and manage ntwk congestion - by ensuring that WAN and LAN infra can support vol of data traff and voice calls.

-shape ntwk traffic - using traff shaping tools

-set traff priorities across the ntwk - marking voice traff as priority and queuing it first.

LLQ provides strict priority qing in conjunction with CBWFQ - it configs the priority status for a class within CBWFQ in wc voice pkts receive priority over all other traff.



Data sent in pkts to remote locs is assembled using a PAD (pkt assem/disassem) into individual pkts of data. Each pkt has unique id marking it as independent and has own dest add.

Fax xmissns are designed to op across a 64kbps pulse code modulated (PCM)-encoded voice circuit. In pkt ntwks, the 64-kbps stream is compressed into much smaller data rate by passing it thru a dig sig proc (DSP). A relay or pass-thru mech has to be in place since the codecs used in the DSP are normally for voice pkts.


- FAX RELAY: involves terminating and xmitting the data on the gtwy.

T.30 fax from PSTN is demod at the sending gtwy. Demod fax is enveloped in pkts, sent over the ntwk and remod into a T.30 fax at the receiving end.

Cisco IOS supports T.38 and Cisco fax relay (proprietary).

- FAX PASS-THRU: invs sending the packet in-band in a Reliable Transport Protocol (RTP) stream. Mod fax info is passed in-band e2e over a voice speech path in IP ntwk.

- STORE-AND-FORWARD FAX:uses T.37 to receive and convert faxes to files. Enables faxes to be delivered via computers rather than fax machines.

also known as voice band data, which refers to the xport of fax or modem signals over a voice channel thru pkt ntwk encoding - min set of coders for VBD is G.711 mu-law and a-law with VAD disabled.

Fax pass thru - simplest tech for sending fax over IP ntwk. Gateway does not distinguish fax calls from voice calls. Fax traff is carried btwn the two endpoints in RTP pkts using uncompressed format resembling the G.711 codec.

A constant 64-kbps (payload) stream is taken plus its IP overhead e2e for the duration of the call. IP overhead for voice traffic is 16-kbps but when switched to pass-thru, the packetization period is reduced from 20ms to 10ms.

Table below compares a G.711 VoIP call that uses 20 ms packetization with a G.711 fax pass-through call that uses 10 ms packetization:

Fax pass-thru is susceptible to jitter, packet loss and latency across the IP ntwk.
The two endpoints need to clock synchronously for this to work predictably.

Performance could be an issue. Redundant recoding (1X or one repeat of the original pkt) is used to mitigate pkt loss, this then doubles the amount of data transferred in each pkt and this in turn imposes a limitation on the number of ports that can run pass-thru at the same time.

One fax pass-thru session with redundancy needs as much bandwidth as two G.711 calls without VAD.

Fax pass-thru does not support the shift from G.Clear to G.711. With G.Clear also configd, the gtwy cannot detect a fax-tone.

Call-control protocols that support fax pass-thru:



Same tech as fax-thru, only modem sig this time.

-does not support switch from G.Clear to G.711
-VAD and echo cancellation need to be disabled.


-represses processing functions like compression, echo cancellation, VAD, high-pass filter.

-issues redundant pkts to protect against pkt drop.

-provides static jitter buffers of 200ms to protect against clock skew.

-Discriminates modem sig from fax sig

-reliably maintains modem connection across the pkt ntwk.


In Cisco Fax Relay mode, gtwys terminate T.30 fax signaling by spoofing a virtual fax machine to the locally attached fax machine. The gtwys use a Cisco proprietary fax relay to communicate btwn themselves.

Unlike fax pass-thru, fax relay demods the fax bits at the local gtwy, sends the info across the voice ntwk using the fax relay protocol, and then remods the bits back to tones at the far gtwy. The fax machines at both ends are not aware of a demod/remod taking place.

Cisco methods for fax relay:

- Cisco Fax Relay -- proprietary and default on most platforms if a fax method is not explicitly configd.

- T.38 Fax Relay: based on T.38 standard, it is real-time fax xmission. Can be configd for H.323, SIP and MGCP.

Features of T.38:

-Fax relay PLC

-MGCP-based fax (T.38) and DTMF relay

-SIP T.38 fax relay

-T.38 fax relay for the T.37/T.38 fax gtwy

-Ti38 for VOIP H.323


Demods a mod sig at one voice gtwy and passes it as pkt data to another voice gtwy, where the sig is remod and sent to a receiving mod. On detection of the mode answer tone, the gtwys switch into mod pass-thru mode and then if the the call menu (CM) sig is detected, the two gtwys switch to mod relay mode.

Two methods are used:

-Modem Pass-through: mod traff is carried btwn the two gtwys in RTP pkts, using uncompressed voice codec, G.711 mu-law, or a-law. Remains susceptible to pkt loss, jitter and latency in the IP ntwk. Pkt redundancy can be used to mitigate effects of pkt loss.

- Modem Relay: The mod sigs are demod at one gtwy, converted to digi form, and carried in Simple Packet Relay Transport (SPRT) protocol, which runs over UDP pkts to the other gtwy, where the mod sig is re-created, re-mod and passed to the receiving modem.

The call starts as a voice-call, switches into mod pass-thru mode, and then into mod relay mode.

Mod relay significantly reduces the effects of pkt loss, latency and jitter on the mod session. Also reduces the amount of bandwidth used.

Features of mod relay:

-modem tone detection and signaling

-relay switchover

-controlled redundancy

-payload redundancy

-dynamic and static jitter buffers

-pkt size

-clock slip buffer management

Gateway Controlled Modem Relay:
new feature supported from Cisco IOS release 12.4(4)T. It is non-negotiated, bearer switched mode for modem transport that does not involve call-agent-assisted negotiation during the call-setup.


The xmitting gtwy is referred to as 'on-ramp gateway', the terminating gtwy as 'off-ramp gtwy'.

-On-ramp faxing: a gateway that handles incoming calls from from a standard fax machine or the PSTN converts a traditional G3 fax into an email message with a TIFF attachment.
Off-ramp faxing: a gtwy that handles outgoing calls from a ntwk to a fax machine or a PSTN converts an email with TIFF attachment to a traditional fax format.

Facilitated with SMTP.

Gateway signaling protocol


-terminating gtwy detects called terminal identification [CED] tone from called fax machine
-terminating gtwy exchanges the codec that was negotiated during the voice call setup for a G.711 codec and turns off echo cancellation and VAD
-this switchover is communicated to the originating gtwy
-originating gtwy allows the fax machs to xfer modem sigs as though they were traversing the PSTN.
-if voice codec is G.711, all that needs to be done is to turn off echo cancellation and VAD


-DSP detects fax or modem : DSP listens for the 2100-Hz CED tone to detect fax or modem on the line.

-internal event alerts the call control stack

-call control stack on the OGW advises DSP to send a named signaling event [NSE] to the TGW informing the TGW of the request to change codec.

-terminating gtwy loads the new codec

NSEs are Cisco proprietary.


When a DSP is put into voice mode at the beginning of a VOIP call, the DSP is informed by the call control stack whether fax relay is supported or not and if it is Cisco fax relay of T.48 fax relay. If CFR is supported:

1. A voip call is established:

-voip call is est as if normal speech.
-call control procedures are followed, after wc human speech is expected to be received and processed.

2. CED tone is detected:

-if fax answer or calling tone [ANSam (modified ANSwer tone) or CED] is heard, the DSP does not interfere with the speech processing.
-the ANSam or CED tone causes a switch to modem pass-thru, if enabled, to allow the tone pass cleanly to the remote fax.

3. A digital info sig [DIS] is sent

4. RTP pkt is received

5. RTP pkts are sent to the TGW


T.38 pass-through and relay uses special protocol enhancements on call control protocols:

1. H.323 T.38 fax relay

What occurs in a H.245 message flow:

-voip call is established - call control procedures are followed, DSp is put into voice mode, human speech is expected to be received and processed.

-CED tone is detected

if ANSam or CED is detected the DSP does not interfere with the speech. ANSam or CED tone causes a switch to modem pass-thru, if enabled, to allow the tone to pass cleanly to the remote fax.

-a DIS message is sent

a normal fax mach after generating a CED or hearing a CNG, sends a DIS msg with the capabilities of the fax mach.

the DSP in the Cisco IOS gw attached to the fax mach that generated the DIS (normally the TGW) detects the HDLC flag sequence at the start of the DIS message initiates fax relay switchover.

DSP also triggers an internal event to notify the call control stack that fax relay switchover is reqd.

the call control stack then instructs the DSP to change the RTP payload type to 96 and to send this payload type to the OGW.

-ModeRequest msg is sent from the detecting TGW to the OGW and the OGW responds with a ModeRequestAck.

-the OGW sends a closeLogicalChannel msg to close its UDP port and the TGW responds with a closeLogicalChannelAck msg while it closes the voip port.

-an openLogicalChannel is sent by the OGW that indicates which UDP port to send the T.38 UDP info on the OGW and the TGW responds with a openLogicalChannelAck msg.

-the TGW sends a closeLogicalChannel msg to close its UDP port and the OGW responds with a closeLogicalChannelAck msg while it closes the voip port.

-an openLogicalChannel is sent by the TGW that indicates which UDP port to send the T.38 UDP stream on the TGW and the OGW responds with a openLogicalChannelAck msg.

-T.38 encoded UDP msgs are sent back and forth

T.38 uses data redundancy to accommodate pkt loss. The level of redundancy can be configd on the IOS GWs.

The T.38 Annex B standard defines the mech that is used to switch over from voice mode to T.38 fax mode during a call.

2. SIP T.38 fax relay

T.38 Annex D procedures are used here for the changeover from voip to the fax mode during a call.

Events during a T.38 fax relay call flow:

-a voip call is est

-CED is detected

-DIS msg is sent

-TGW detects a fax V.21 flag sequence and sends an INVITE msg with T.38 details in the Session Description Protocol [SDP] field to the OGW or the SIP proxy server.

-the INVITE msg is received by the OGW and sends back a 200 OK msg.

-TGW acknowledges the 200 OK msg and sens an ACK directly to the OGW.

-OGW starts sending T.38 UDP pkts instead of voip UDP pkts across the same ports.

-at the end of the fax xmission another INVITE msg can be sent to return to the voip mode.

3. MGCP T.38 fax relay

Two modes of implementation:

1. Gateway-controlled mode:

2. Call agent-controlled mode:

Call flow for MGCP T.38 fax relay:

-a call is est as a voice call

-gateways' capabilities are advertised in SDP exchange during connection establishment

-support for T.38 fax relay is checked

-connection is reverted to voice call upon completion

A fax relay MGCP event allows the gw to notify the call agent of the status(start, stop or failure) of T.38 processing for the connection. this event is sent in both modes of implementation.

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Tuesday, January 26, 2010

Jots for today



-multiple independent sites, with own call-processing agent cluster connected to an IP WAN

A distributed call-processing site may consist of:

-single site with its own call-processing agent (may be CUCM or CUCM Express, or other IP PBX)

-a centralized call-processing site and all of its associated remote sites.

-legacy PBX with VOIP gateway

IP WAN interconnects all the distributed call-processing sites - PSTN serves as a backup connection between sites in case of IP WAN failure. A site connected only thru the PSTN is a standalone site and is not covered by dist call-proc mod.


-leased lines

-Frame Relay


-ATM and Frame Relay service internetworking (SIW)

-Multiprotocol label switching (MPLS) VPN

-Voice and video Enabled VPN (V3PN) IPSec protocol

Multisite dist call proc allows each site to be independent.
IP WAN failure or insufficient bandwidth wiil not cause a site to lose call-proc service or functionality. CUCM simply send all calls between the sites across the PSTN.



Max 30,000 Skinny CCP (SCCP) or Session IP (SIP) IP phones or SCCP video endpoints per cluster

-MGCP GATEWAYS or H.323 devices

1,100 per CUCM cluster


for external call

-DSP resources

for conferencing, transcoding and MTP

-Voice mail and unified messaging

-legacy PBX and voice-mail systems

-H.323 clients, multipoint control units, H.323/H.320 gateways

Multipoint control unit resources

required for multipoint video-conferencing.

H.323/H.320 video gateways: needed to comm with H.323 video conferencing devices on the ISDN network.

-High-bandwidth audio:

-High-bandwidth video

-Min of 768kbps or higher WAN link speeds - video not recomm for WAN links with speed lower than 768kbps.

-CUCM locations and automated alternate routing (AAR) - CUCM provides CAC.


-Cost savings on using IP WAN for calls between sites

-Use of IP WAN for tail-end hop-off (TEHO) - IP WAN is used to bypass toll charges by routing calls thru remote gateways closer to the PSTN number dialed.

- Max util of avail bandwidth by allowing the voice traffic to share the IP WAN with other types of traffic

-No loss of functionality during an IP WAN failure

-Scalability to hundreds of sites


A multisite dist call proc deployment is a superset of the Single-site deployment and the multisite WAN with centralized call proc - follow the best practises of both.


Gatekeeper or SIP proxy servers are among key elements in this model

-they provide dial plan resolution, with gatekeeper providing CAC.

-a gatekeeper is an H.323 device that provides CAC and E.164 dial plan resolution.


-use a Cisco IOS gatekeeper

-use HSRP gatekeeper pairs, gatekeeper clustering and alternate gatekeeper support

-size platforms appropriately

-use only one type of codec on the WAN - because H.323 does not allow for Layer 2, IP, UDP, or RTP header overhead in the bandwidth request.

Gatekeeper networks can scale to hundreds of sites.

SIP devices provide provide resolution of E.164 numbers as well as SIP Uniform Resource Identifiers (URIs) to enable endpoints place calls to each other. CUCM supports the use of E.164 numbers only


-provide adequate redundancy for the SIP proxies

-ensure that the SIP proxies have the capcity for the call rate and number of calls reqd in the network.



-recommended size: up to 240 phones

-small remote sites

-capacity depends on IOS

2. CUCM:

-recommended size: 50 to 30,000 phones

-small to large sites depnding on size of the CUC

-supports centralized or dist

3. Legacy PBX with VOIP gateway:

-recommended size: depends on PBX

-number of IP WAN calls and functionality depends on the PBX-to-VOIP gateway protocol and the gateway


A single CUCM cluster and its subscriber servers are split across multiple sites connected via a QoS-enabled WAN


1. local failover deployment model

CUCM subscriber and backup server have to be at the same site with no WAN in between. Ideal for 2 or 4 sites with CUCM.

2. remote failover deployment model

Backup servers are allowed to be deployed over the WAN. can have up to 8 sites with CUCM subscribers being backed up CUCM subscribers at another site.

Might need higher bandwidth because of a large amount of intracluster traffic

A combo of both deployments can also be used.


-single point of administration for users for all sites within the cluster

-feature transparency

-shared line appearances

-extension mobility within the cluster

-unified dial plan

Ideal as a disaster recovery plan

-useful for customers who require more functionality than the limited feature set offered by the SRST

-allows remote offices to support more Cisco IP phones than SRST does in the event connection to CUCM is lost.


The Intra-Cluster Communication Signaling (ICCS) between CUCM servers consists of many traffic types. The ICCS traffic types are classified as either priority or best effort.

Priority ICCS traffic is marked with IP Precedence 3 (DSCP 24 or PHB CS3). Best-effort ICCS traffic is marked with IP Precedence 0 (DSCP 0 or PHB BE).

The following design guidelines apply to the indicated WAN characteristics:


The maximum one-way delay between any UCM servers for all priority ICCS
traffic should not exceed 20 ms, or 40 ms round-trip time (RTT). Delay for other
ICCS traffic should be kept reasonable to provide timely database access.
Propagation delay between two sites introduces 6 microseconds per kilometer without
any other network delays being considered. This equates to a theoretical maximum
distance of approximately 3000 km for 20 ms delay or approximately 1860


Jitter is the varying delay that packets incur through the network because of
processing, queue, buffer, congestion, or path variation delay. Jitter for the IP
Precedence 3 ICCS traffic must be minimized using QoS features.

-packet loss and error

The network should be engineered to provide sufficient prioritized
bandwidth for all ICCS traffic, especially the priority ICCS traffic. Standard
QoS mechanisms must be implemented to avoid congestion and packet loss. If packets
are lost due to line errors or other “real world” conditions, an ICCS packet will be
retransmitted because it uses the TCP protocol for reliable transmission. The retransmission might result in a call being delayed during setup, disconnect
(teardown), or other supplementary services during the call. Some packet-loss conditions could result in a lost call, but this scenario should be no more likely than errors occurring on a T1 or E1, which affect calls via a trunk to the PSTN/ISDN


Provision the correct amount of bandwidth between each server for the
expected call volume, type of devices, and number of devices. This bandwidth is in
addition to any other bandwidth for other applications sharing the network, including
voice and video traffic between the sites. The bandwidth provisioned must have
QoS enabled to provide the prioritization and scheduling for the different classes of
traffic. The general rule for bandwidth is to overprovision and undersubscribe.


The network infrastructure relies on QoS engineering to provide consistent and
predictable end-to-end levels of service for traffic.

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Monday, January 25, 2010

Still on Today's reading: Single-Site and Multi-Site Deployment Models

The Single-Site Model consists of:

-call-processing agent cluster located at a single site or campus with no telephony services provided over an IP WAN.
-all CUCMs, Apps, Data-stream Protocol (DSP) are in the same location.

-multiple clusters inside a LAN or MAN can be implemented and connected through inter-cluster trunks if more IP phones needed here.


-Single CUCM cluster
-Max 30,000 Signaling Connection Control Part [SCCP] or Serial Interface Proc (SIP) IP phones or SCCP video endpoints per cluster.
-max 1,100 H.323 devices per CUCM cluster
-PSTN for all calls outside the site.
-DSP resources for conferencing, transcoding and Media termination Point [MTP]
-voice mail, unified messaging, Cisco Unified Presence, audio and video comps.
-capability to integrate with legacy PBX and voice-mail systems

-H.323 clients, multipoint control units, H.323/H.320 gateways: if gatekeeper is required these must register with a Cisco IOS gateway release 12.3(8)T or later. CUCM then uses an H.323 trunk to integrate with the gatekeeper and provide call routing and bandwidth management services for the H.323 devices registered to it.

-Multipoint control unit resources: req for multipoint video-conferencing. May be either SCCP or H.323 or both.

-H.323/H.320 video gateways:needed to comm with H.323 video conferencing devices on the ISDN network.

-High-bandwidth audio:

-High-bandwidth video


-ease of deployment
-a common infrastructure for a converged solution
-simplified dial plan
-no required transcoding resources(single high-bandwidth codec in use)

-significant cost benefits
-allows CUC to take adv of many IP-based apps on the enterprise.
-allows each site to be self-contained - no dependency for service or loss of call-processing in the event of IP WAN failure or insufficient bandwidth.


-provide a highly available, fault-tolerant infrastructure
-know the calling patterns for your enterprise
-use G.711 codec for all endpoints - this eliminates the consumption of DSP resources for transcoding
-use SIP, SRST and MGCP gateways for PSTN - this simplifies the dial plan config.
-implement the recommended network infrastructure.


Consists of {

-a single call-processing agent cluster that provides services for many remote sites and uses the IP WAN to transport CUC traffic between the sites.

The IP WAN also carries call-control signaling between the central site and remote sites


-leased lines
-Frame Relay
-ATM and Frame Relay service internetworking (SIW)
-Multiprotocol label switching (MPLS) VPN
-Voice and video Enabled VPN (V3PN) IPSec protocol

Routers at WAN edges req QoS mechs (priority queuing and traffic shaping) to protect voice traffic from dta traffic across the WAN where bandwidth is typically scarce.

Call Admission Control (CAC) is also needed to avoid oversubscribing the WAN links with voice traffic and deteriorating quality of established calls.

When the IP WAN is down, a variety of Cisco gateways can provide the remote sites with PSTN access.


-single CUCM cluster
-max 30,000 SCCP or SIP IP phones or SIP video endpoints per cluster.
-max 1000 locations per CUCM cluster
-max 1,100 H.323 devices or 1,100 MGCP gateways per CUCM cluster.

-PSTN for all external calls.
-DSP resources for conferencing, transcoding and Media termination Point [MTP]
-voice mail, unified messaging, Cisco Unified Presence, audio and video comps.
-capability to integrate with legacy PBX and voice-mail systems

-H.323 clients, multipoint control units, H.323/H.320 gateways: if gatekeeper is required these must register with a Cisco IOS gateway release 12.3(8)T or later. CUCM then uses an H.323 trunk to integrate with the gatekeeper and provide call routing and bandwidth management services for the H.323 devices registered to it.

-Multipoint control unit resources: req for multipoint video-conferencing. May be either SCCP or H.323 or both.

-H.323/H.320 video gateways:needed to comm with H.323 video conferencing devices on the ISDN network.

-High-bandwidth audio:

-High-bandwidth video

-Min of 768kbps or higher WAN link speeds - video not recomm for WAN links with speed lower than 768kbps.

-CUCM locations and automated alternate routing (AAR) - CUCM provides CAC.

-must have SRST versions 4.0 and higher as they support video

-CUCM Express versions 4.0 and higher - maybe used in place of SRST routing.\

-Cisco Unity Server - CUCM Express maybe be integrated with the Unity Server in the branch office or remote site.


-Minimize delay between CUCM and remote locations to reduce voice cut-through delays (clippings). 150 ms maximum one-way is Cisco's recomm.

-Use the locations mech in CUCM to provide CAC into and out of remote branches.

-Number of branch IP phones and line appearances that are supported in the SRST mode at each remote site depends on the branch router platform, the amount of installed mem and IOS releases.

SRST on a Cisco IOS gateway supports up to 720 phones, whereas the CUCM Express in SRST mode supports 240 phones.

The choice of whether to adopt a centralized call-processing model or a distributed call-processing model depends on:

-IP WAN bandwidth or delay limitations

-criticality of the voice network

-Feature set needs


-Ease of management


In the event a dist call-proc mod is chosen, the choices include installing the CUCM cluster at each site or running CUCM Express at remote sites.

-another design guideline is to use these features for call processing survivability in the event of WAN failure at remote sites:

a. for SCCP phones use CUCM in a Cisco IOS gateway or CUCM Express in SRST mode.

b. for SIP phones, use SIP SRST

c. for MGCP phones, use MGCP gateway fallback

They can all reside with each other on a Cisco IOS gateway.

-Hot Standby Router Protocol (HSRP) may also be used to provide backups for gateways and gatekeepers in a VoIP environment.

**CAC is used to avoid oversubscribing the WAN links with voice traffic and deteriorating the quality of established calls.

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Sunday, January 24, 2010

Today's jottings


Actual voice data is transported using media transmission protocols like RTP and RTCP.


- defines format for delivering audio and video over the Internt
- delivers audio and video streams over a network


- provides OOB control info for an RTP flow.

2 Protocols which enhances the use of RTP:

- cRTP which compresses IP/UDP/RTP headers on low speed serial links.

- SRTP which provides encryption, message auth and integrity, and replay protection to RTP data.

RTP and RTCP are both built on UDP.


- e2e network transport functions like payload-type ID, sequence numbering, time stamping, delivery monitoring.

- runs on top of UDP in order to use UDP's multiplexing and checksum services. This is because it has no standard TCP or UDP port on which it communicates (though it is sometimes config to use ports 16384 or 32767). UDP comms must be done via an even port and the next higher odd port is used for RTCP.

- can carry any data with real-time characteristics. Call setup and teardown is usually performed by SIP. It is difficult for RTP to traverse firewalls since it uses dynamic port config. It works well with queuing to prioritize voice traffic over other traffic.

- designed for multicast sessions. It defines roles os sender/receiver and also roles of translator/mixer to support multicast reqs. But it is often used for unicast sessions.

- includes various services like payload-type ID, sequence numbering, time stamping, delivery monitoring.

RTP is used:

- with RTSP (real time streaming protocol) in streaming media systems
- wih H.323 or SIP in video conferencing and push-to-talk systems

What makes RTP so critical a component for VOIP is the fact that it enables destination devices to reorder and retime voice packets before they are played out to users.

An RTP header contains a time stamp and a sequence number, which allows the receiving device to buffer to remove jitter and latency by synchronising the packets to play back a continuous stream of sound.

Sequence number is only used for ordering the packets. There is no request by RTP for retransmission if a packet is lost.


-sender report packet
-receiver report packet
-source description RTCP pkt
-goodbye RTCP pkt
-app-specific RTCP pkt


-provides oob control info for an RTP flow

it does not transport any data itslef, but provides feedback on the QoS being provided by RTP.

-used for QoS reporting

Monitors quality of data distribution and provides control info

-provides mech for hosts involved in an RTP session
-provides a time stamp based on data pkt sampling
-provides separate flow from RTP for transport use by UDP


RTP is divided into 2 portions, data and header.

The data portion is a thin protocol and provides support for real-time properties of applications, such as continuous media, including timing reconstruction, loss detection and content identification.

The header is made up of iP/UDP/RTP segments which is 20/8/12 bytes which equals 40 bytes. This is simply too large and it is inefficient to send this without compressing it. The header compression feature compresses the 40 bytes to approx 2 to 4 bytes.

RTP payload is 20 to 150 bytes for audio apps that use compressed payloads.

RTP header compression should not be used on any high-speed interfaces - anything over T1 speed.

SRTP - secure RTP standardizes the utilization of only a single cipher, AES. which can be used in 2 cipher modes and turns the original AES cipher block into a stream cipher.

The two cipher modes:

1. Segmented integer counter (SIC) mode: allows random access to any block and is essential for RTP traffic running over an unreliable network with possible loss of pkts. AES running in this mode is the default encryption algorithm, with a default encrytion key length of 128 bits and a default session salt key of 112 bits.

2. f8 mode: a variation of output feedback mode. Default values above applies.

SRTP also allows users to disable the cipher using 'NULL cipher'.


Gateways provide a way for IP telephony networks to connect to the pSTN, PBX, etc.

A voice gateway functions as a translator between different network. Coverts protocols between terminals such as H.323 and PBX. Connects company networks to PSTN, PBX, etc.



Two subcategories here:


Connects IP telephony ntwk to POTS.
Provides FXS ports to connect analog tels, interactive voice response (IVR)systems, fax machs, PBX systems and voice-mail systems.


Connects IP telephony ntwk to the PSTN central office (CO)or a PBX.
Provide FXO ports for the PSTN or PBX access, and E&M ports for analog trunk connection to legacy PBX.
Analog direct inward dialing [DID] is also available for PSTN connectivity.


Cisco access digital trunk gateways connect an IP telephony ntwk to the PSTN or PBX via digital trunks like PRI common channel signaling (CCS), BRI and T1 or E1 channel associated signaling (CAS)


-support gateway protocols
-provide advanced gateway functionality
-work with redundant CUCM
-enable call survivability
-provide Q signaling [QSIG]support
-provide fax and modem support

MGCP might be the preferred option to H.323 for gateway config due to its simpler config. It works well with older Cisco IOS versions because it supports call survivability during a CUCM failover from a primary to a secondary CUCM.

H.323 on the other hand may be preferred for its advanced features.


-standard for integrating voice-mail systems to PBX or Centrex systems.
-connecting to a voice-mail system via smdi using either analog fxs or T1 PRi requires either SCCP or MGCP since H.323 does not identify the specific line that is being used by a group of ports.


1. Dual Tone Multifrequency (DTMF) relay capabilities:

-each digit dialed with tone dialing is assigned a unique pair of frequencies
-DTMF tones are separated from the voice bearer stream and sent as signaling indications via the gateway protocol.

2. Supplementary services support:

-provide user functions such as hold, transfer and conferencing.

3. Must have the ability to REHOME to a secondary CUCM in the event that the primary fails.

4. Must enable call survivability in the CUCM.

- the voice-gateway preserves the RTP bearer stream (the voice conversation) between two IP endpoints when the CUCM to which the endpoints are registered is no longer accessible.

5. Should provide QSIG support (which is becoming the standard in Europe and North America)

- with QSIG, the cisco voice pkt appears to PBXs as a distributed transit PBX that can establish calls to any PBX or other telephony endpoint served by a Cisco gateway, including non-QSIG gateways.

6. Should also provide fax and modem support

- the fax image is converted from an analog signal and is transmitted as digital data over the pkt ntwk.

Gateways are normally deployed as edge devices on a network.

Some enterprise gateway hardware:

1. Cisco 2800 Series Integrated Services Router: four models-

-Cisco 2801
-Cisco 2811
-Cisco 2821
-Cisco 2851

-provides up to five times the overall performance
-up to 10 times the security and voice performance
-embedded service options
-dramatically increased slot performance and density

-can deliver simultaneous, high-quality, wire-speed services up to multiple T1/E1 or xDSL connections.

The routers offer:

-embedded encryption acceleration
-voice digital signal processor(DSP) slots

-intrusion prevention systems (IPS) and firewall functions

-optional integrated call processing and voice-mail support

-high-density interfaces for wide range of wired and wireless connectivity requirements.

-sufficient performance and slot density for future network expansion reqs and advanced apps.

2. Cisco 3800 Series Integrated Services Router

also feature:

-embedded security processing
-significant performance and memory enhancements
-new high-density interfaces which deliver the performance, availability and reliability reqd to scale mission critical security, IP telephony, business video, network analysis and web apps in the most demanding enterprise environments
-deliver multiple concurrent services at wire-speed T3/E3 rates

-are designed to embed and integrate security and voice processing with advanced wired and wireless services for rapid deployment of new apps

-support bandwidth reqs for multiple FE interfaces per slot, TDM interconnections and fully integrated power dist to modules supporting 802.3af PoE.

-supports the existing portolio of modular interfaces.

3. Cisco Catalyst 6500 Series Switches

-high-performance and feature rich platforms that can be used for voice gtws by installing a Cisco Communications Media Module (CMM)

-CMM is a line card for the Cat 6500 series. It provides flexible and high-density VoIP gtwy and media services.

-The Series can handle many digital trunk interfaces eg, up to 144 T1/E1 conns by using 8 CMMs with 18 ports each.


1. Cisco 1751-V Modular Access Router

-supports multiservice integration of video, voice, data and fax traffic.
-offers many WAN-acces and voice-interface options, VoIP high-performance routing with bandwidth management, inter-VLAN routing, and VPN access with a firewall options.

2. Cisco 1760-V Modular Access Router

-19 inch rack mount solution geared towards small-medium businesses
-supports multiservice integration of video, voice, data and fax traffic.
-offers many WAN-acces and voice-interface options, VoIP high-performance routing with bandwidth management, inter-VLAN routing, and VPN access with a firewall options.

3. Cisco 2600XM Series Multiservice Routers
4. Cisco 3600 Series Multiservice Routers
5. Cisco 3700 Series Multiservice Routers

All of them are either EOS or almost-EOS.


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Saturday, January 23, 2010

What is VOIP?


Voice over IP converts voice into digital signals that travel over the internet.

TDM - means traditional time-division multiplexing which is what traditional PSTN networks use. A lot of bandwidth is lost here when there is no voice traffic.

PACKET TELEPHONY - This is packet-based voip transmission and is able to multiplex voice traffic side by side with data traffic - more efficient.

-ip phones
-multipoint control unit
-call agent
-video conferencing station



- the ability to generate and exchange control information that will be used to establish, maintain and release connections between two endpoints

Voice signalling requires the ability to provide supervisory address and alerting functionality between nodes.

PSTN uses SS7 to transport control messages. SS7 uses OOB signalling(exchange of call control info in a separate dedicated channel)


- the ability to query databases - this is needed to detremine if a call can be placed or not. (Billing info, CNAM delivery, toll-free dtabase services, calling card sevices, etc)


- carry voice calls
-PSTN network: connect/disconnect messages carried by SS7
-IP network: carried by SIP, H.323, H.248, MGCP


- provide coding/decoding translation between analog and digital facilities (e.g. G.729)

1. P2P:
-end devices of gateways contain the intelligence to initiate/terminate calls interprete call control messages.
-SIP, H.323

- no call control intel but can send/receive event notifications to the server (call agent./)
- H.248, SCCP, MGCP


1. H.323
Specifies the components/protocols/procedures that provide multimedia services over packet networks including IP networks - real-time audio/video/data comms.

Also defines end2end call-signaling.

Five protocols specified by H.323 are:

a. H.225 Call Signaling - used to establish connection between two H.323 endpoints.
b. H.225 registration, admission and status (RAS) - lies between endpoints (terminals and gateways) and gatekeepers. RAS is used to perform registration, admission control, bandwidth changes and status and disengage procedures between endpoints and gatekeepers.
c. H.245 Control Signaling - used to exchange e2e control messages governing the op of the H.323 endpoint. The control messages carry info relating to

- capabilities exchange
- opening and closing of logical channels used to carry media streams
- flow-control messages
- general commands and indicators

c. Audio Codec - encodes the audio signal from the mic for transmission on the transmitting H.323 terminal and decodes the received audio code sent from the receiving H.323 terminal to the speaker.

Audio is the minimum standard provided by H.323, hence all H.323 terminals must have at least one audio codec support as specified by the ITU-T G.711 recomm (audio codec at 64kbps).

Some other audio codec recomms that may also be supported:

- G.722 (64, 56, and 48 kbps)
- G.723.1 (5.3 and 6.3 kbps)
- G.728 (16kbps)
- G.729 (8 kbps)

D. Video Codecs - Like the audio one but having to do with video.
H.323 must conform with the ITU-T H.261 recommendation.

Remember, H.323 is a p2p protocol so H.323 gateways are not registered with CUCM as an endpoint, it needs an IP add to be configured in CUCM for comm to take place.

The necessary gateway configs are complex - need to define dial plans and route patterns directly to the gateway. H.323 does all the signaling between the CUCM and the gateway

Some H.323 capable devices are:

- Cisco VG224 Analog Phone Gateway
- Cisco 2600XM Series Multiservice Routers
- Cisco 2800 Series Integrated Services Routers
- Cisco 3700 Series Multiservice Access Routers
- Cisco 3800 Series Integrated Services Routers

Defines a method for PSTN gateway control or thin device control.
Controls VOIP gateways connected to external call control devices (call agents).
Provides signaling ability for cheap edge devices that may not have full voice-signaling protocol, such as H.323, implemented.

MGCP is a client/server protocol and is built on centralized control archi.

All dial plan info reside on a separate CALL AGENT, which controls ports on the gateway and performs call control.

The gateway does media translation between the PSTN and VOIP networks for external calls. CUCM functions as CALL AGENT in a Cisco network.

MGCP is plain-text.

MGCP allows complete control of the dial plan from CUCM which it also gives per port control of connections to the PSTN, legacy private branch exchange (PBX), voice-mail systems and POTS, etc. This control is implemented by series of plain-text commands sent between CUCM and the gateway over UDP port 2427. the gateway must have CUCM support.

A PRI and BRI backhaul is an internal interface between the call agent eg CUCM and the Cisco gateways.

A PRI backhaul forwards PRI Layer 3 (Q.931) signaling info via a TCP con.

A call agent must always be available for a MGCP gateway, which is itself relatively easy to configure since the call agent has all the call-routing intelligence.

Cisco MGCP gateways can use SRST and MGCP fallback to allow the H.323 prot to take over and provide local call routing in the absence of a CUCM. Dial peers must then be configured on the gateway for use by H.323.

MGCP is relatively easy to config due to its client/server archi. The dial plan and route patterns are defined directly on the CUCM within the cluster.

MGCP is used to manage the gateway. All ISDN Layer 3 (Q.931)info is backhauled to the CUCM. The ISDN Layer 2 (Q.921) info is terminated on the gateway.

3. SIP

Is P2P
Specifies commands to initiate and tear down calls.
Features security, proxy, tcp, udp.
Works with SAP and SDP to to provide announcements and info about multicast sessions to users on a network.
Defines e2e call-signaling between devices.
ASCII Text-based
Much like HTTP (same transaction request/response model and similar header/response codes.)

Since it is p2p, CUCM does not control SIP devices and SIP devices do not register with CUCM.

Only IP add is used to confirm communication between CUCM and SIP gateway (much like H.323).

The necessary gateway configs are complex - need to define dial plans and route patterns directly to the gateway. SIP does all the signaling between the CUCM and the gateway. The ISDN prots Q.921 and Q.931 are used only on the ISDN link to the PSTN - same as H.323


Used between CUCM and Cisco VOIP phones.
End stations using SCCP are called SKINNY clients (consume less processing overhead).

It is a stimulus prot - any event triggers messages to the CUCM, which then send it detailed instructions on what to do about the events.

Cisco devices using SCCP can coexist with H.323 terminals.


To configure voice on a data network, there has to be low delay,minimal jitter and minimal packet loss. Bandwidth requirements must be met, QoS must be configured to minimize jitter and loss of voice packets.

1. Latency

- Increase Bandwidth
- Choose diff codec type
- Fragment data packets
- Prioritize voice packets

2. Jitter

- use dejitter buffers

3. Bandwidth

- Resolve bandwidth issues by calculating bandwidth requirements including payload , overhead and data.

4. Packet Loss

- design network to minimize congestion
- prioritize voice packets
- use codecs to minimize small amounts of packet loss

5. Reliability

- provide redundancy for hardware, links, and power (UPS)
perform proactive network management

6. Security

- Secure network infrastructure
- secure call-processing systems
- secure endpoints
- secure apps

My Cisco folly

Have being Cisco CCNA certified for sometime. Still waiting to land that dream Cisco job! While am waiting I have decided to chase my next cert - Cisco CCVP. With the hoopla about convergence and its importance in VOice transmission over IP, I have decided to go this way for now. Already have my Cisco QoS exam done. Next on the list is the CVOICE v6.0!