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May 07 2013

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642-437 CVOICE v8.0 Passed – Have Some Notes

So last Friday I went in after ~2 1/2 weeks of study and passed the CVoice 8.0 exam. So I’m 1/5 of the way to my CCNP Voice. After a project wraps up at work, I’ll start in on CIPT 1. Here is a raw dump of the notes I took during my study that seem to cover the majority of what I was hit with on the test.

If you’re curious about the test check out the Cisco Learning Network.

https://learningnetwork.cisco.com/community/certifications/ccvp

https://learningnetwork.cisco.com/community/certifications/ccvp/cvoicev8?tab=1

 

Layer 2 Headers Size [bytes]
802.3 Ethernet 18
802.1Q Ethernet 18+4
PPP 6 to 9
MLPPP w/ Interleaving 13
Frame Relay 6
Frame Relay w/ FRF.12 8

Layer 3 Overhead (IP/UDP/RTP) is 40 bytes

 

Codec Packetization Period Codec Payload Packet Per Second
G.711 – default 20ms 160 bytes 50
G.729 – default 20ms 20 bytes 50
G.711 – alternate 30ms 240 bytes 33
G.729 – alternate 30ms 30 bytes 33

 

8900 / 9900 phones are Cisco SIP fully featured
Signaling
Call setup / tear down
ISDN / SS7 E&M -Analog
H323 – suite of protocols – peer to peer
MGCP – Client / Server – CUCM
SIP – peer to peer
SCCP – endpoints and call processing agent – client / server
Phones / PVDMs

H323 – Layer 5 of OSI session

Debug ISDN q931 – Layer 3 – Call Setup / Tear down
Debug ISDN q921 – Layer 2 –

MGCP Benefits is a centralized dial plan
H323 is a de-centralized dial plan

Residential gateway configured for POTS
Trunking gateway is ISDN

SIP – Uses ASCII IETF RFC

SCCP – endpoint control protocol

5 Components of a Dial Plan
o Endpoint Addressing
o Route / Path Selection
o Digit Manipulation
o Class of Restriction
o Call Coverage
Dial-peer is the cornerstone of the dial plan
CUCME – add phones & users v8.x
Gateway – Routes the calls IOS v15.x

Deployment models
Single – Site
CUCM
IP WAN data only
PSTN used for calls
H.323 or MGCP for gateways
Uses single best quality G.711 /722 codec
Provides DSP for conferencing / MTP
Offers services
HSRP for gateway HA
GLBP – recommended for VoIP
QoS – End to End
Security
CUBE

Multi-site deployment
CUCM at HQ
Phones at branch office need registration and Call Processing
WAN down CFUR – Call Forwarding Un-registered
AAR -Automatic Alternative Routing – used if WAN BW is exceeded
CAC – Call Admission Control – limit of number of call between the site (limited by BW)
Apply QoS
At remote sites use SRST ( can be coupled with CME for more features)

Multi-site distributed
Multiple clusters at multiple sites
Uses a gatekeeper to centralize the dial-plan
IP WAN carries inter-cluster calls
Gatekeeper for CAC
DSP use for resources

Clustering over IP WAN
80ms< RTT
AAR used if no BW
CAC
EF for VoIP traffic

Gateway HW Platforms
ISR G2 IOS G2 15.x
Version 8 cucm

Gateway Operation modes
Voice Switching – PBX to PSTN – analog to analog
VoIP gateway – PSTN to IP – analog to digital SIP
CUBE- IP to IP gateway – digital to digital

Call Legs
POTS Dial-peer – analog
VoIP Dial-peer – digital / IP
Dial-peer is like a static route – hop by hop route
Every dial-peer match creates a call leg

Signaling protocol analog
SS7, ISDN, QSIG

CUBE
Proxies signaling
Address hiding
NAT

HOW DIAL PEERS FUNCTION
Each matched dial-peer is a call leg
An addressable call endpoint
Establish logical connections to other dial-peers
Uses two call legs

Types:
Pots – analog
VoIP – IP / DNS name
MMoIP – Fax to SMTP email store and forward
POTS Dial-peer
Pointed to a port the phone is connected to
VoIP dial-peer
Pointed to an IP / DNS

MGCP does not use dial-peers

We can set the following under dial-peers
Codec
VAD
DTMF relay

Destination-pattern 2…. Like ip route x.x.x.x x.x.x.x y.y.y.y
Port 0/1/1

Dial-peers closest match wins eg. Dial 2001 choice is 2…. Or 20.. / 20.. Wins as it’s a more exact match

Each dial-peer starts with 0 preference
Two dial-peers with the same destination pattern (preference would be the tie-breaker)
Dial-peer 0 – default dial peer

POTS Dial-peer
Defaults to discarding all match digits (digit stripping)
Forward-digits all
Needs a port

Use of String Matching
Call Number – DNIS
Calling Number – ANI Automatic Number Identification
Incoming called-number matches called number in inbound dial-peer
Answer-address – matches calling number in inbound dial-peer

Matching Char
\+ – matches the plus sign
+ – means the preceding digit occurred one or more times
^ – start of string
$ – end of string
? – zero or more times must press ctrl-v to disable help to put in ?
T – inter-digit timeout – 10 seconds CME / T302 15 seconds CUCM
\ – followed by character matches it.
[] – indicates a range
() – indicates a pattern

Matching on inbound DP
Use either the DNIS (DID) or ANI (caller ID)
Port

T1 – 23 channels bearer
E1 – 30 channels bearer
Each with 1 D channel for setup

1. Incoming called-number
2. Answer-address
3. Destination-pattern
4. Port (POTS DP ONLY)
5. Default dial-peer

ANSWER ADDRESS
Useful for matching a geographical region
Incoming Called Number
Recommended for most configurations
Destination Pattern
Outbound dial-peer

Default Dial-peer VoIP – Inbound Only
DP 0
G.729 or G.711
No DTMF relay
IP precedence 0
VAD enabled – No good for fax / modem
No RSVP – IntServ QoS
Fax-rate voice
Default Dial-peer POTS – Inbound Only
No Applications
No DID

Avoid using DP 0
Non-default parameters will fail
Many errors are due to codec, VAD an dtmf-relay misconfig when using DP0
AS5350xx – requires inbound DP

Direct Inward Dial
Two-stage dialing
Call arrives pots, gw provides Dial tone and takes more digits
One-stage dialing
Call arrives pots, no dial-tone called gateway receives entire called number.

DID not supported on FXS/FXO/E&M
DID available on FXS-DID and digital circuits

Voice Ports
Analog FXS / Analog T1 (RBS)
FXO – POTS LINE
E&M – Used for Trunk circuits
Digital T1 ISDN
CAMA Trunk – for 911
CAS – Channel Associated Signaling
CCS – Common Channel Signaling

Analog Signaling
FXO/FXS
Loop Start
Ground Start
Address signaling
Pulse
DTMF
Information signaling
Progress tones – dial tone / busy etc.

Digital Voice Ports
T1 CAS – RBS – Every 6th frame is robbed for signaling
E1 TDM
ISDN
BRI – 2B 64k bonded for 128k 1D Channel
T1 PRI – 1.472 Mb/s 23 B / 1 D
E1 PRI – 1.920 Mb/s 30 B / 1 D
Uses CCS – Common channel (dedicated D)
NFAS – Non-facilited associated signaling – shares D channel across multiple PRI
MGCP – does not support NFAS

QSIG – interpreter between different PBXs
MWI
Call forwarding

CCS – D channel signaling
23 Bearer channels

Layer 2 – Q921
Layer 3 – Q931
Cptone – dial tone
Copand-type a-law mu-law

Isdn overlap-receiving – change en block to digit by digit analysis
isdn incoming-voice voice – Incoming voice calls bypass modems and handled as voice calls.

Deployment models
• Single Site
• Mult-site Centralized Call Processing
• Multi-site Distributed Call Processing
• Cluster Over WAN
o <80ms RTT
o QoS

Echo Cancellation
Echo is a signal that leaks from RX to TX path
Talker Echo – When you hear yourself (most common) – leak from rx to tx at remote end
Listener Echo – When you hear an echo when you – leak from rx to tx at both ends
Causes:
Impedance mis-match at the two wire to four wire hybrid connections.

G.168 Echo Canceller
Enabled by default
Faces into the PSTN
Echo canceller coverage is the length of time that is stored in memory.

ERL (echo return loss) – reduction of returning echo (larger is better)
Use output attenuation and input gain to tune ERL to at least 6 dB

Show call active voice – call detail units
Csim start xxxx

DSP
P-1-144 Codec List
Concentrate on G711 and G729 for testing
MOS – Mean Opinion Score
G729B vs G729A
A – less complex and less quality
B- Adds VAD and CNG
G.711 – 33/50pps Payload 240/160 bytes
G729 – 33/50/pps Payload 30/20 bytes
G711 – MOS 4.3
G729 – MOS 3.92

G711 – 20
G729 – 20

BW per call =(Voice Payload + Layer3 overhead + layer 2)*packets per second * 8 bits
Needed for CAC
LLQ
Test will ask about Ethernet and Frame Relay layer 2 overhead
Layer 3 + G711 over frame relay, 50 pps

DSP Function
Voice Termination
Voice termination
Echo Cancelation / VAD / Jitter Management
MTP
Call hold
transform between a-law and mu-law
Transcoding
Convert from one codec to another
Conferencing

Flex – subject to oversubscription
PVDM3 allows sharing between conf, mtp, transcoding
PVDM2 allows sharing between mtp,transcoding

Configuring DSPs
High
Medium
Flex
Secure – Encrypted RTP

Module 2
RTP Types
RTP – Normal
RTCP – RTP over TCP – Used for monitoring gathering statistics
cRTP – compresses IP/UDP/RTP headers on low-speed links <768kb/s
SRTP – encryption of RTP, message authentication, integrity and replay protection
RTP
Picks random ports from UDP port 16384 – 32767
Payload type identification – codec type can change during use
Sequence number
Measures delay/jitter
RTCP
Used for monitoring RTP stream
Out of band network monitoring
Compressed RTP
Compresses the RTP header, IP/UDP/RTP = 40bytes compress it down to 2 or 4 w/ CRC bytes.
On high speed links processing overhead not justified
One-way audio if CRTP on one side not the other
Secure RTP
Encryption
Message Integrity
Message authentication
Replay protection

VoIP Considerations
Firewalling need ports opened
Don’t IPSEC SRTP packets.

VAD
When silence detected packets discarded until speech is heard.

H323 Signaling
Based on Q931 structure
Vendor-neutral
Peer-to-peer
Self-sufficient dial plan per gateway
Translations defined per gateway
Support for ISDN NFAS
Enhanced fax support and call preservation
H323 GW
Translation between audio, video, data
Conversion between call setup signals and procedures
Conversation between communication control signals and procedures
H323 GK
Centralizes distributed h323 dial plan
RAS
Used between GW and GK
Fast Start
H323 Fast Start is version 2
Combination of TCS / OLC into one packet
Early Media
Offers streaming of MoH Announcements before call setup is complete
VoIP Dial Pear
By default is an H323 dial peer
H323 Tuning
H323 session transport tcp/udp
H323 source IP address
H323 timers

Recommended to bind H323 to a loopback interface

Voice class h323 fancy_phones
H225 timeout tcp establish 3
H225 timeout setup
Dial-peer voice 100 pots
Voice-class h323 fancy_phones

H323 Resource thresholding is Disabled
H.225 RAS message RAI
RAI goes from gateway to the gatekeeper
SIP
Creates modifies and terminates multimedia sessions
RTP,RTCP,HTTP,SDP,DNS,SAP,MGCP,RTSP
Support on Cisco Gateways
Peer-to-Peer
Uses ASCII text-based messages
SDP establishes the lowest common service level
Can do mid-call changes, codec changes, transfer, hold

SDP
Carries the media
Transport protocol RTP/UDP/IP….
The format of media (codecs)
Delayed offer is part of SIPv1
Early offer is part of SIPv2

MGCP
Centralized device control
Allows for remote control of various devices
Stimulus protocol
Endpoints and gateways cannot function alone
Uses UDP
Simplified IOS configuration

TCP 2428 – Terminates data link layer back to CUCM for PRI Backhaul
MGCP + ISDN PRI/T1 or E1
Q931 setup happens on CUCM when PRI is backhauled
MGCP port 2427 defaults to UDP

Show mgcp
Show ccm-manager – see registration status of MGCP

Audio Quality
Delay – the time it takes for the signal to propagate from one end to the other.
Jitter: variation in the arrive of voice packets
Fidelity: audio accuracy
Echo: usually due to impedance mismatch
Side tone: allows speakers to hear their own voice
Background noise – low volume noise heard at the far end

Delay Sources
DSP delay [FIXED]
Serialization delay [FIXED]
Propagation delay [FIXED]

Compression [VARIABLE]
Shaping [VARIABLE]
Network [VARIABLE]
Forwarding / Processing [VARIABLE]
Queuing delay [VARIABLE]

SLA
150< ms one way is acceptable delay
One way Jitter should be less than 30ms
Packet loss should be 1% or less
BW Requirements
Voice media 17 – 106kb/s
SCCP – 150b/s

QoS Mechanisms
Header Compression
CRTP
FRF.12
Splits larger packets into smaller fragments – specific to sub-interfaces
IP RTP Priority and Frame Relay IP RTP Priority
Prioritization of voice media traffic
LLQ / Strict Priority Queue (CLI – priority 256) (queuing)
Voice prioritization over data packets
MLPPP
Link aggregation
Resource Reservation Protocol RSVP
Method for CAC
PSTN fallback
Backup over PSTN if network service below required level

QoS does not give more BW just better manages it via:
Prioritization
Shaping

Methods to transmit fax and modem over IP networks
Fax-relay – demodulated forwarded in packets and re-modulated
Fax pass-through – passed in-band end to end over VoIP path
Store-and-forward T.37 – Breaks fax into distinct sending and receiving processes.

Cisco IOS supports T.38 and Cisco fax relay protocols
Fax pass-through
G.711
No VAD
No Echo canceller
H.323 / SIP / MGCP Supported

Fax Relay
Carries signal out of band via SPRT (simple packet relay transport)
Two Modes Cisco fax relay and T.38
H.323 / SIP / MGCP supported

Max transfer speed of 33.6kb/s

Store and forward T.37
On-ramp – Fax terminates at gateway is converted to TIFF to sent via SMTP email
Off-ramp – Email sent to gateway with TIFF and is modulated and sent out to the PSTN

MGCP has two fax relay modes
Gateway controlled
Call Agent controlled

DTMF is distorted by lower bandwidth codecs – DTMF Relay is the solution
Tones are sent out of band
DTMF Relay Methods H.323
Cisco proprietary – In-band Identified with payload type 121
H.245 signal – Out-of-band DTMF sent via H.245 instead of RTP
H.245 alphanumeric – Out-of-Band DTMF sent via H.245 does not send tone length (requires H.323v2)
RTP Named Telephony Events (NTEs) -In-band special NTE RTP formats used to signify digit type
None – In-band, tones left in the audio stream (default setting)

DTMF Relay Methods SIP
Sip Notify – out-of-band forwarded via SIP NOTIFY messages
RTP Named Telephony Events (NTE)- In-band special NTE RTP formats used to signify digit type
None – In-band tones are left in the audio stream. (Default Setting)

DTMF Relay Methods MGCP
Cisco Proprietary – DSPs code the DTMF digits differently so they can be identified.
RTP NSE – Uses RFC 2833 signaling method
RTP NTE – Gateway controlled (gateways exchange SDP to determine capabilities) and call agent controlled (CUCM instructs gateway to process DTMF)
Out-of-band – Sends tones as signals to the call agent over the control channel.
None – In-band tones are left in the audio stream. (Default Setting)

Max-conn {number} – specifies the total concurrent (in / out) connections per dial-peer (default no limit)

QoS
QoS.pdf – Cisco best practice on QoS
Converged networks
Time sensitive
Sensitive to packet loss
Quality Issues
Lack of bandwidth
End to end delay (fixed / variable)
Variation of delay (jitter)
Packet loss
L2- 802.1Q – Frame tagging – Class of Service
L2 – 802.1p – 3bit of the 4B CRC header
L3 – TOS – Type of Service

Old – IP Precedence – 8 queues
New – DSCP – 64 queues
WFQ – Weighted Fair Queuing – Moves small packets to front of line (default)
CBWFQ – Class based weighted fair queuing – you determine who goes to the front of the line
1. Classify
2. Mark
3. Set Policy
Ways to increase BW
Upgrade the link
Forward the important packets first
Compress the payload of L2 frames (takes time)
Compress IP Packet headers
End to end delay = the sum of all the propagation, processing and queuing to get across the network.

Processing Delay
Queuing Delay
Serialization Delay
Propagation Delay

CBLLQ – recommended queuing – Class Based Low Latency Queuing

Tail Drop – occurs when the output queue is full
WRED – Weighted Random Early Detection – tail drop algorithm

QoS – The ability for the network to provide better “special” service for a set of users or applications (voice)

Voice QoS requirements (one way)
Latency < 150ms
Jitter < 30ms
Loss <1%

Priority in config means LLQ

Three Models
Best Effort – no qos applied
IntServ -Applications signal the network the require special QoS. This is RSVP
DiffServ -Network recognizes classes that require special QoS. Hop by Hop – DSCP Values

DSCP – Value in the IP header TOS byte
BA – Behavior aggregate – grouping of same DSCP packets
Per-hop behavior (PHB) – the QoS treatment applied by a node

CoS – Class of Service 802.1P L2 marking
CoS to DSCP Mapping – Converts L2 marking to L3 Marking

PHB
Default – FIFO
EF – Expedited Forwarding
Ensures a minimum departure
Guarantees bandwidth
Polices bandwidth
AF – Assured Forwarding
Guarantees bandwidth
Allows access to extra bandwidth, if available
Class-selector (CS) – IP Precedence PHB

Classification –
Marking
Congestion Management – LLQ
Congestion Avoidance – WRED
Policing and Shaping –
Link Efficiency – Compression, link fragmentation, interleaving

Policing – Very severe, dropping traffic that exceeds limit / or re-mark
Shaping – When limit is reached packet is re-queued / buffered

Any Link <768kb/s
Should run cRTP and LFI

TOS – L3 marking – DSCP/IP Precedence
COS – L2 marking – COS bit

Class-map- define traffic
Policy-map – What to do with defined traffic
Contains class-map
Service-policy – apply to interface
Applies Service-policy

Match all – must match all conditions to be matched – Default
Match any – must match any condition to be matched

Class-map {match-all | match-any} VoIP
Match ip precedence 5
Match ip dscp ef cs5

CEF must be enabled on the interface before class-based packet marking feature can be used

Trust Boundaries
At the phone endpoint and access layer is optimal
Distribution layer acceptable
LLDP – Link Layer Discovery Protocol (industry standard CDP)

MLS qos trust device cisco-phone

*Cos to IP Precedence / DSCP map may change depending on switch.*

show mls qos maps cos-dscp – displays the L2 to L3 mapping at the switch

RTP Header compression
Compresses 40byte header down to 2 or 4 with checksum

Compression header ip rtp / tcp – done in a policy map for a specific class

Shape software interfaces to match HW interfaces.

Traffic policing drops / remarks – Policing in and out (Inbound / Outbound)
Traffic shaping buffering minimizes TCP re-transmits – shaping out only – Outbound only

Conform / exceed / violate
Drop / set (remark) / transmit

Configure shaper in b/s or percentage

CBWFQ
Each class has a reserved queue
Each class can perform WRED
Each class gets more than reserved BW when there is no congestion

LLQ separate priority queue running parallel to CBWFQ

Show policy-map interface f0/1 – shows the applied policy and what packets are hitting what queue

QPM – GUI tool to see QoS

Cisco Auto QoS for enterprise – Support on routers only

Calculating Call BW
BW = (codec payload + L3 overhead + L2 overhead) * packet rate * 8
E.g. A G.711 call at a 20ms packetization period over Ethernet
(160+40+18)*50*8=87.2K per call

Call Manager Express
Call Processing
Signaling and device control
Dial Plan Administration
Phone Feature Administration
Directory services
Direct access to gateway features and modules

Up to 365 users for CME
H323 / SIP / SCCP support no MGCP
Virtual dial-peers created for ephone-dn’s

Endpoint requirements
CDP
DHCP
MAC Address
TFTP
Power brick or PoE
Cisco in-line power – eol
802.3af
SEP – Selcius Ethernet Phone

PoE Providers
PoE switched
Power Injectors
Wall Power AC-Power

Show int f0/1 switchport
ip-helper propagates the broadcast into unicast for DHCP

Telephony-service
Ip source-address
Max-ephone 50
Max-dn 100

Ip dhcp pool blah
Network 10.0.0.0 255.255.255.0
Default-router 10.0.0.1
Option 150 ip 10.0.0.1

< 768k Recommendations
cRTP – compresses 40byte header to 2 or 4 byte
LFI – Link Fragmentation and Interleaving
LLQ / Strict Priority Queue
Class-map
Classify Traffic
Policy-map
Mark Traffic
L2 – Cos 802.1p
L3 – DSCP / IP Precedence
Service-policy
Apply Policy via service-policy on Interface
Auto-QoS Enterprise (routers only)
Auto QoS Discovery – 1 – 7 days
Vlan 8021p – marks COS on access only ports for voice
Support SIP phones CME – voice register pool
By Default DN’s are single-line
Oct-line on SCCP Phones only
SCCP Phones CME default to G.711
SIP Phones CME default to G.729

Cisco IP Phone
Optimize for low BW = G.729
Reset = hard reboot
Restart = soft reboot

Telephony-service (all phones)
Restart
Reset
Show ephone – registered SCCP phones
Show voice register all – registered sip phones

A Number plan must include your internal and local numbering plan. NAMP
NAMP

5 Dial Plan Components
1. End Point Addressing
2. Route / Path Selection
3. Digit Manipulation
4. Class of Service (COR)
5. Call Coverage

Scalable Number plan
Hierarchical
Summarization
Simplicity in provisioning
Reduction in post-dial delay
Availability for fault tolerance
Conformance to public standards

Intr-asite and inter-site
For example users can use 4digit to all extensions in the enterprise

Site codes used to overcome overlapping / poorly structured dial plan
Auto-attendant used when no operator and no DID numbers

AAR – Used when not enough bandwidth on WAN
ISDN TON – Type of Number
Subscriber
National
International
Unknown

Digit Collection and Consumption
Num-exp – global
En Block – all digits are sent in one block
Digit by digit – each one is individual
POTS Dial-peers digit strip
No digit-strip
VoIP Dial-peers don’t digit strip
All things being equal random selection – preference is tie breaker

Voice Translation rules
1. Create Rule
a. Up to 15 rules within a rule
2. Create Profile
a. Give it descriptive name
3. Apply Profile – usually to a dial-peer

Rules are assigned to called, calling , redirect-called

Test voice translation-rule

Inbound Dial-peer matching
1. Incoming called number
2. Answer address
3. Destination pattern
4. Port (pots only)
5. Default DP
Outbound dial-peer match
1. Destination pattern
2. Preference wins
a. If equal preference then random

Best practice
Inbound dial-peer
Incoming called-number .
Direct-inward-dial

H225 timeout tcp establish 3 <– failover to another node if no TCP established

COR – Class of Restriction
Define the destination that the user is allowed to dial
IOS gateways use COR
Rely on proper call routing
Requires high dial-peer granularity
Different destinations defined in separate dial peers

COR lists applied to dial-peers
Incoming COR list – Key
Outgoing COR list – Lock

COR list are assigned to E-phone for CME
COR list assigned under SRST

Call-manager-fallback
Cor incoming executive 1 2000 – 2100

Gatekeeper
Area code -> IP Address
Call Routing
Address Resolution
CAC (optional)

Zone Local – my local table
Zone Remote – remote table of area code to ip

Gatekeepers can be logically separated from h323 gateways
Gatekeeper Mandatory functions
Address resolution
Admission control
Zone management

RAS UDP messages from GW to GK
RAI – resource availability indicator – from GW to GK
GW registers with H323 ID or E164 Address
H323v1 every 30s sent full registration
H323v2 starts with full registration
Then lightweight registration
LRQ – location request sent in the following ways
Sequential – slower but less signaling
Blast- faster routing more signaling

Zone
Zone prefixes – DNs or area codes

Technology prefix
Used to group GWs via X# code regardless of zone data.
1# default
GK CAC is Codec times 2 – provided to:
CUCM
CME
H323 Devices
Number of call *Codec BW *2

Gatekeeper should be fully configured before no shut

CUBE
H323 to SIP
H323 to H323
SIP to SIP
External Connections
VoIP carriers
Interconnect with other voice and video networks
Integrate internet VoIP and Video over IP users
Relevant features
Address hiding
Security
Video integration
CAC

Solves interoperability issues with different signaling protocols
Flow-through – default
Signaling and media through cube
Hides IP addresses
Enabled tighter security
Flow-around
Media streams flow between endpoints
Only on H323 to H323 and SIP to SIP

Destination-pattern
Media flow-around
Codec transparent
Session-target

Call start interwork – allows fast to slow start h323

CUBE uses standard IOS with RSVP support
Allow-connections h323 to sip
Allow-connections sip to h323
*Uni-directional rules for bi-directional traffic both statements needed*

REVIEW
Voice translation
Create Rule
Create Profile
Apply profile to voice-port

Telephony-service
Dialplan-pattern matches incoming number tells it extension length and the pattern to start with

Show dialplan number
Show voice translation-rule
Test voice translation-rule

COR Setup
Dial-peer cor custom
Name 911
Name local
Name national
Name intl

Dial-peer cor list outbound-employee
Member 911
Member local
Member national

Dial-peer cor list outbound-executive
Member 911
Member local
Member national
Member intl

Dial-peer cor list call-911
Member 911

Dial-peer cor list call-local
Member local

Dial-peer cor list call-national
Member national

Dial-peer cor list call-intl
Member intl

Ephone-dn 30
Number 1000
Description Employee
Cor incoming outbound-employee

Ephone-dn 31
Number 1001
Description Executive
Cor incoming outbound-executive
k
Dial-peer voice 911 pots
Destination-pattern 911
Forward-digits 3
Cor list outgoing call-911
Port 0/0:23

Dial-peer voice 100 pots
Destination-pattern 9[2-9]..[2-9]……
Cor list outgoing call-local
Port 0/0:23

Dial-peer voice 101 pots
Destination-pattern 91[2-9]..[2-9]……
Prefix 1
Cor list outgoing call-national
Port 0/0:23

Dial-peer voice 111 pots
Destination-patter 9011T
Prefix 011
Cor list outgoing call-intl
Port 0/0:23

Permanent link to this article: http://tripplehelix.net/642-437-cvoice-v8-0-passed-have-some-notes/

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