//Paging. Paging is the ability to make an announcement on speaker phone for a group of people. Could be used for a fire senerio.
Dial the extension to make the announcement. First create the groups for your teams
CCP -> Configuration -> Unified Comms -> Telephone Features -> Paging Numbers -> Add
Then create the groups of group below to specify all people.
CCP -> Configuration -> Unified Comms -> Telephone Features -> Paging groups -> Add
//After hours restrictions. Stop calls after a certain time.
CCP -> Configuration -> Unified Comms -> Telephone Features -> After-Hour tollbar
//Single number reach. Essentially means that if an extension is dialed, then other devices ring aswell (mobiles/home phone)
Configuration -> Unified Communications -> Users + Phones and Extensions -> extensions -> Advanced -> Single number reach
—Sample size.
Is the amount of audio included in each packet. Configured under the dial peer which is your
settings to other VOIP Gateway devices. default sample size 20ms for G.711, higher ms size, means less
overhead however more delay.
Sample size is calculated by below formalae
Some selections of byte size include
80,160,240 (Default is 160)
—Codecs. Ways to compress audio
G.711 =64 kb uncompressed Audio MOS 4.1
G.729 =8 kb compressed MOS 3.92 //very resource intensive on DSPs
G.729A=8 kb compressed MOS 3.7 //less heavy on DSPs
G.726 =32kb Compressed MOS 3.82
G.728 =16kb compressed MOS 3.61
ILBC (Industry Standard) Compressed = 15.2 Kb MOS 4.1
—VAD
Stands for Voice activity detection.
Suppresses the silence in each converstation. Saves about 35% of bandwidth.
—Compressed RTP
Strips off the headers and can save up to 40% of bandwith.
Very processer intensive task, more than a T1 connection then you may hammer your router.
—RTP
Uses UDP on a random EVEN port between 16 364 and 32767
—RTCP
Uses UDP on a random ODD port between 16 384 32767
—Calculating bandwidth required for VOIP
bytes per sample = (Sample size X Codec Bandwidth) /8
//set on codec in dial peer //Codec type sets this.
240 = (SS x 64,000) / 8
x8 x8
1920 = SS X 64,000
/64000 /64000
.03 = Sample size in seconds. which is 30 ms.
Multiply by 1000 to make it in ms.
//Add in network overhead @ layer 2
Ethernet = add 18 Bytes of overhead
Frame Relay = 4-6 Bytes of overhead
PPP/MLPP = add 4 bytes of overhead
//Add network layer 3+4 information + 40 bytes total
IP = 12 bytes
UDP = 8 bytes
RTP = 12 Bytes
//Add in Bonus overhead
GRE / L2TP = 24 bytes
MPLS = 4 bytes
IPSEC: 50 -75 bytes
//Do the calculation
bytes per sample = (Codec bytes + L2 + L3 + L4) = 218 bytes
Calculate samples per second = 1000 / SS in seconds (30ms in above example) = 33.33
Calculate packets per second = 33.33 X 218 = 7265.94bytes
Calculate bytes to bits = 7265.94 x 8 =58127.52
Calculate bits to Kbits = 56.76515625 kbits per second per user.
—DSPs (Digital signaling processors)
These are used to offload processing from the routers CPU to the DSP module as the router is not designed to handle the bellow tasks.
-Conferencing
-Transcoding (Converting between u-law and a-law)
-Encoding (Using a Codec to convert Audio)
-MTP (Call waitng from PSTN lines)
Use a DSP calucator on the internet to evaluate how many DSP’s are required
—Router Gateways
convert between different call types, EI between VOIP and PABX. or between external calls and
internal calls. Can have 2 different types of ports, analogue (FX0 ports) which take 1 call per phone connection or digital ports which take multiple calls.
—FXS ports.(anologue = 1 call)
FXS cards are used to add anologue phone line ports to your routers for internal use
Maybe you have an anologue phone in the lift/ a fax machine (alternative is IP FAX machines)
ATA186/188 can be used to get a single analogue port from an ethernet jack
—FXO ports (anologue = 1 call)
Add phone lines to your router to get external phone lines.
Can link to telecommunications provider or to an exsisting PABX solution.
Known as trunking, to Teleco or PABX.
—VWIC (digital supports multiple calls on 1 line)
Can connect a WAN port or act as a voice TRUNK for PSTN connections.
Trunk can connect to an exsisting PABX or to a Telco provider.
T1 + E1 CAS (OR RBS) supports 24(t1)/(e1)30 calls per line. steals bits for signal line degrades quality. signalling bit is stolen from every 6th frame to carry signalling information. (CAN LOOSE FEATURES LIKE CALLER ID)
T1 + E1 CCS (Primary rate interface) – supports 23(T1)/30(E1) line calls, uses a deadicated line for signalling, works with more features. (WILL NOT LOOSE A LINE FOR E1, NO LOSS OF FEATURES)
BRI (Basic Rate Interface) – Supports 2 lines + 1 for signalling.
—Gateway protocols
Used for communication between voice gateways.
H.323 – old protocol, heavy overhead chuncky. P2P each router has its own config on codecs and
how to reach extensions.
MGCP – cisco propitary some other vendors support, this is a server-client model where routers
download information from a server to know what codecs to use, and how to reach each extension
MGCP can require registration of cucm server
SIP – industry standard, still new features arent 100% compiatable between different vendors products. designed to replace H.323