-----------------------------------------------------------------------------------------------------------------------------
CIPT Notes on Weak
Areas
·
EF
– 46 – 101|110
·
The
# instructs interdigit timeouts to stop
o
You
need two route patterns identical if you want the user to be able to use the #
symbol in CUCM
·
Digits
sent one by one to CUCM begin digit analysis immediartely
o
Even
going off hook digit analysis starts (for PLAR situations)
·
SIP
en block sends the entire string at once in an INVITE
·
KPML
enables digit by digit for SIP
·
SIP
dial rules are processed inside the SIP phone so it knows what it can and
cannot dial
o
If
matched, a single INVITE with the digits are sent en bloc
o
If
more digits are needed, KPML can take over
·
ISDN
PRIs and trunks can be configured to allow overlap sending and receiving
o
This
permits digits to be sent or received one by one over the PRI
o
Need
to check the “Allow Overlap Sending” check box in the route pattern
configuration page
o
Set
the “OverlapReceivingFlagfForPRI in the service parameters to True
·
SCCP
o
Type
A are digit by digit only
o
Type
B is en bloc or digit by digit
o
Can
disabled en bloc if needed
·
Translation
patterns always have urgent priority enabled and cannot be disabled
o
En
bloc dialing will make urgent priority to not be considered since it is not
digit by digit
·
Cannot
reference a single device as a member of both a route group and a direct reference
to a route pattern
·
Route
groups are a list of devices that share the same requirements for digit
manipulation
o
i.e.
PSTN gateways all within a route group
o
Circular
selection of gateways within a route group can be used to load balance
o
Top
down can be used for priority as well as providing a backup gateway in the
event the primary fails
·
Local
route groups decouple the selection of a PSTN gateway or trunk for off-net dialing
from route patterns that are used to access the gateway.
·
Standard
Local Route Groups can be added to each route list entry once
·
Local
Route Group within the device pool so it uses the local group for that device
pool
·
Within
the Route List, you can setup digit manipulation per route group
·
Class
of Service, also called Calling Privileges define entries of a call-routing
table that can be accessed by an endpoint that performs a call-routing request
o
Controls
telephony charges
o
Restricts
international calls
o
Can
be used to route calls differently with the same number per endpoint
o
Route
different based on time of day
·
Partitions
are defined as a group of numbers with similar reachability characteristics
·
CSS’s
define what partitions are accessible from a particular device
·
CTI
ports the device CSS comes before the line CSS
o
This
is different than the standard device in which the line is concatenated on top
of the device
CUCM Media Resource
Notes:
·
Transcoding – requires DSPs on the gateway or
blade
·
Voice termination – Requires hardware for TDM to
VoIP
·
Audio Conference Bridge – Requires either
software or hardware resources
·
MTP – Can either be software based CUCM or hardware
on gateway
·
Annunciator – Software only
·
MOH – Software only unless in SRST mode on
gateway
You need to start the Cisco IP Voice Media Streaming
Application to activate the following resources:
·
Audio Conference Bridge
·
MTP
·
Annunciator
·
MOH
o
SRST at remote sites if you want this to be
hardware based, otherwise will not work if connected to CUCM out of SRST
·
Signaling between hardware media resources and
CUCM use SCCP
·
All media resources register with CUCM
·
Audio streams are always terminated by media
resources
·
No direct IP to IP phone audio streams present
when media resources are involved
Conference Resources:
·
Software bridge runs on one or more CUCM servers
in a cluster
·
Audio Streams go between:
o
IP phone to conference bridge
o
Gateway to conference bridge for those PSTN guys
o
CUCM does not distinguish between software and
hardware based conference bridges when it processes a conference allocation
request
Transcoding:
·
Same concept as conference bridges in terms of
signaling
·
Audio Streams between IP Phones and transcoder
and between application server and transcoder
·
Runs on the IOS router and registers via SCCP
·
Converts G.711 to G.729 and vice versa as well
as other codecs in the same fashion
·
Hardware based resources for transcoding
residing on the IOS gateway (PVDM)
MTP Signaling and Audio Steams
·
MTP bridges two media streams and allows them to
be set up and torn down independently
·
Audio streams between the phones and the MTP
·
Same signaling concept as before. Between IP Phone and CUCM and between MTP and
CUCM
·
MTPs can be used to translate between two
incompatible media streams
o
Can be used to synchronize clocking and enable
supplementary services like hold/resume
MTP Types
·
Three types of MTPs
o
Software MTP on CUCM
§
Provides G.711 mu-law to a-law and packetization
conversion
·
i.e. 20ms to 30ms packetization sample size
o
Software MTP on IOS routers
§
No DSPs needed
§
Same codec and packetization on both call legs
·
THEY MUST BE IDENTICLE!!!
§
For functions like RSVP or CUBE media
flow-through configurations
§
CUCM sees every
Cisco IOS software MTP as a hardware MTP!!!
§
Basically supplementary services like
hold/resume only without DSPs
§
Enable this by using the maximum session
software <n> command
o
Hardware MTPs
§
DSPs (PVDM modules) required
§
Use of same audio codec but different
packetization on different call legs possible
·
Config Example on IOS gateway:
sccp ccm group 1
associate ccm 1
priority 1
associate profile 1
register IOS-HW-MTP
associate profile 2
register IOS-SW-MTP
!
dspfarm profile 1 mtp
codec pass-through
codec g711ulaw
maximum sessions
hardware 2
associate application
SCCP
!
dspfarm profile 2 mtp
codec g711ulaw
codec pass-through
maximum sessions
software 100
associate application SCCP
MTP Functions and Requirements
·
Inserting DTMF
o
Supported on all MTP types
·
G.711u to a conversion
o
CUCM soft MTP only
·
Sample size conversion
o
CUCM soft and IOS DSP MTPs only
o
IOS Software will not do this, only hold/resume
services
·
RSVP signalling
o
Only Cisco IOS soft and hard MTPs
·
Provide H.323v1 supplementary serbices
o
All MTPs available can do this
Annunciator
·
A software function of the Cisco IP Voice Media
Streaming Application
·
Streams spoken messages or various call-progress
tones from the system to a user
·
Can send multiple one-way RTP streams to devices
like:
o
IP Phones
o
Gateways
·
Uses SCCP messages to establish the RTP stream
·
Device MUST support SCCP to use this feature
·
Support localization and can be replaced with
.wav files
·
Can support G.711u and a-law
o
Also supports G.729
o
Wideband
o
No additional resources required to do any of
this!
·
Audio stream is one way only from annunciator to
the IP Phone
·
Signaling is between:
o
IP Phone to CUCM
o
Annunciator to CUCM
o
Between Gateway and CUCM
MOH Signaling and Audio Streams
·
Provides music to users on hold, transferred,
parked, or added to an Ad Hoc Conference
·
Simple but need to know unicast and multicast
traffic
·
Need to also know MOH Call flows, configuration
options, and server behavior / requirements
·
Audio streams between the IP Phones and the MOH
Server and between the gateway and the MOH Server
·
Signalling between IP Phone and CUCM
o
Between MOH Server and CUCM and between gateway
and CUCM
Conference Bridge Overview
·
CUCM supports both hard and soft conference
bridges
·
Soft bridges only support single-mode
conferences in the G.711 codec
·
Some hardware-based conferences support
mixed-mode conferences with different codecs from different participants
o
i.e. low bit rate(LBR), G.729, GSM, or G.723
·
This mixed mode converts these codecs to G.711
·
Then converted back to the user original codec
outbound
·
Some hardware conference bridges still only
support G.711
Software Audio Conference Bridge
·
Part of the Cisco IP Voice Media Streaming App
service
·
Limitations:
o
Unity cast audio streams only
o
Any combo of G.711u-law, G.711a-law, or wideband
audio streams may be connected
o
Max audio streams per server is 128
§
If the Cisco IP Voice Media Streaming app runs
on the same server as CUCM service, software conferences should not exceed 48
participants
·
Ad Hoc Conferences
o
3 minimum participants
o
64 max
o
4 default participants
·
Meet-Me
o
1 minimum
o
128 max
o
4 default participants
Hardware Audio Conference Bridge
·
All conference bridges under CUCM control use
SCCP to communicate to CUCM
·
CUCM does not distinguish between soft and hard
conference resources
·
Secure conferencing reduces session capacity by
half with G.711
·
Number of conferees per session is the same as
non-secure
Built-in conference bridge resources
·
IP Phones with built-in conference bridge
resources allow three way conferences
·
Invoked via Barge feature only
·
G.711 only
·
Advanced Ad Hoc lets any participant add and
remove others
·
Link multiple Ad Hoc conferences together
Conference Bridge Service Parameters
·
Accessed in the Cisco IP Voice Media Streaming
App
·
Call Count for max participants
o
Can be 0 to 256, default is 48
·
Run flag needs to be set to true for software
conference bridging
CUCM Service Parameters related to Conferencing
·
Can suppress MOH in Conference bridges
·
Drop Ad Hoc Conference
o
Never (default)
o
When Conference Controller Leaves
o
When No On-Net parties remain in the conference
·
Advanced Ad Hoc conference enabled (false by
default)
o
Can let other users add/remove people
o
This is done with the ConfList or RmLstC softkey
o
Must allow the advanced features within the
advanced Ad Hoc to join conferences
·
Non-linear Ad Hoc conference linking enabled
(false)
o
Allows more than two Ad Hoc conferences to be
joined
·
Max Ad Hoc Conference (4)
o
Default is 3-64, this determines how many people
can be in a conference
·
Max Meet Me Conference (4)
o
Max number of particpants
MOH Media Resources
·
CUCM has an integrated MOH software server
·
External media-streaming servers can be used
·
Supports multicast and unicast
MOH Sources
·
One fixed source that uses a Cisco MOH USB audio
sound card
·
50 audio file sources
·
MOH Audio File management converts the audio
file
·
Codecs for MOH:
o
G.711
o
G.729
o
Wideband
§
G.729 is for speech and reduces music quality
·
Format for MOH files
o
16-bit PCM .wav file
o
Stereo of mono
o
Sample rates of 48,32,`6, or 8 kHz
·
Fixed audio can be transcoded in real time by
CUCM
o
Transcoded into G.711, 729a, and wideband
o
Only audio source that can do this is fixed
since a sound card is involved
Unicast MOH
·
Stream sent directly from server to endpoint
·
Separate source stream for each new connection
·
Point to point one way RTP audio stream
·
Can strain the network in terms of bandwidth
·
Good for networks without multicast
Multicast MOH
·
Point to multipont stream from the server to a
multicast group IP Address
·
Conserves resources since multiple endpoints use
the same audio source stream
·
Must support multicast on the network
·
IP address starts at 239.1.1.1 to
239.255.255.255 for multicast IPs
·
Every different audio source will increment the
IP address by one
MOH Audio Source Selection
·
Holder and a Holdee
o
Holder is the person placing the call on hold
o
Holdee is the person being placed on hold
·
MOH Stream sent to an endpoint is a used is a combo of the User Hold MOH
Audio Source that is configured for the holder and the prioritized list of MOH
resources in the MRGL for the holdee
·
Basically, the source determines what file is
used while the list determines what server is used
·
All files need to be on all MOH servers if
multiple exist or nothing will be heard if a file is missing that is being used
MOH Configuration
·
Plan MOH capacity
·
Configure MOH audio sources
·
Check MOH Server configuration
·
Check MOH Service Parameters
·
Configure Multicast MOH if needed or wanted
o
Configure audio sources for the multicast
o
Configure the MOH server for multicast
o
Implement a MRGL where MOH Multicast is enabled
·
It is imperative that MRGLs be configured with MRGs
o
The MRG multicast checkbox needs to be checked
or multicast will not be utilized when the resources are requested
MOH Server Capacity Planning
·
Max of 51 unique sources per cluster
·
Default of 250 unicast sessions per server
·
Each multicast MOH audio source must be counted
as 2 MOH streams
·
Max of 204 multicast streams
o
51 sources x 4 codec types
·
Maximum half-duplex streams determine the max
number of devices that can be placed on unicast MOH
o
Default value is 250
·
Maximum multicast connections determines the
maximum number of devices that can be placed on multicast MOH.
o
Default value is 30,000
MOH Service Parameter Verification
·
Supported codecs
o
G.711u/a
o
G.729a
o
Wideband
·
QoS for MOH
·
Packet size for codecs is 20ms
·
Cisco CUCM Service
o
Supress MOH in conference = true
o
Default Network Hold MOH Audio Source ID = 1
o
Default User Hold MOH Audio Source ID = 1
o
Duplex Stream Enabled = False
MOH Server Config (again)
·
Enable multicast
·
Verify IP
·
Verify base port number
·
Set max hops TTL for MOH packets
o
This caps out the hops MOH can traverse, if set
to 0 it stays in the subnet only!
·
Increment based on IP not port, this helps save
the firewall and network congestion
Annunciator Features and Capacities
·
G.711, G.729, and wideband without the need for transcoding
·
The following features REQUIRE an annunciator
o
MLPP (Multi Level Precedence and Preemption)
§
Plays a
message based on preemption of call or not
o
Integration via SIP Trunk
§
SIP endpoints can generate and send DTMF in-band
in RTP
§
SCCP cannot do this so MTPs are used in
conjunction with the annunciator
§
Call progress and DTMF tones are sent to SCCP
phones in this manner
o
IOS Gateways and intercluster trunks
§
These devices require support for the call
progress tone (ringback tone)
o
System Messages
o
Conferencing
§
Plays a barge-in tone for entrance and leaving
Annunciator Performance
·
Annunciator running co-resident with CUCM can
support 48 simultaneous annunciator streams
o
That number is the max recommended
·
Annunciator without the CCM service running can
run 255 simultaneous streams
·
10MB/s or lower servers should be dropped to 24
simultaneous streams
·
High performance servers can run 400
simultaneous streams
Media Resource Access Control Overview
·
By default, all media resources are in a NULL
MRG
o
Load balanced
·
Hardware conference resources are preferred due
to mixed mode support
o
Less load on CUCM too
·
Media Resource Manager (MRM) manages all media
resources within a cluster
·
Use media resource management to let hardware
and software resource co-exist
·
Allows them to be used with different priorities
·
Can share and access resources within the
cluster
·
Can load distribute within a group of similar
media resources
·
Media access control bundles MRGs in MRGLs which
prioritizes media resources
·
This can also deny certain users access to
hardware resources if you place them in a different MRGL with no access to
hardware resources
Media Resource Design
·
MRGs define logical groupings of media servers
·
Associate MRGs with their geographical location
is possible
·
Can control unicast and multicast in this
fashion too for MoH
·
MRGLs are a prioritized list of MRGs
·
When a device needs a media resource it will
check its own MRGL first then it will go to the default MRGL
o
Default MRGL contains all media resources in a
default MRG that have not been specifically assigned to another custom MRG
§
For example, site 1 has site 1 MRGL and site 2
has site 2 MRGL
§
Site 3 does not have a MRGL but is co-located
near site 1
§
When site 1 needs resources and is tapped out of
its own media resources it can use site 3s media resources because they were
not specifically assigned to any MRG/MRGL
Intelligent Bridge Selection
·
CUCM can intelligently use video conference
resources over audio if two or more devices are video capable
o
If only one is video and the other is audio, it
will choose an audio bridge instead so that the video bridge resource can be
left for others to use
·
Audio will be used in place of video bridges if
there are no video bridges available
o
This is for Ad Hoc only
·
To configure the parameters of Intelligent
Bridge Selection, go to CUCM service parameters and set the intelligent bridge
selection parameters
·
Encrypted audio or non-encrypted video
·
Set the number of video capable devices that
need to be present to use a video bridge
·
Can set video CFB when available over an audio
if needed
Media Access Control Configuration
·
Create MRGs of resources
·
Create MRGLs
·
Assign MRGLs to phones
·
Note that MRG order in MRGL are not relevant
unless multiple MRGs contain the same type of resource. For example, two hardware MRGs with
conference resources would be used in the order provided in the MRGL
o
The same if vice versa, if you need an
annunciator but the first MRG in the MRGL does not have one, it skips to the
second one immediately
No comments:
Post a Comment