Multi-Tech Systems Voice Over IP A Primer For Resellers Instructions Manual

							Copyright © 2003 Multi-Tech Systems, Inc.  All rights reserved.9
Latency is defi ned as the average “travel” time it takes for a packet to pass through the network, from 
source to destination. The average time varies according to the amount of traffi c being transmitted and 
the  bandwidth  available  at  that  given  moment.  If  the  traffi c  is  greater  than  the  bandwidth  available, 
packet delivery will be delayed.
MultiVOIP  deals  with  the  latency  issue  in  a  private  network  as  well  as  over  the  public  Internet.  In  a 
private network, when network traffi c is at peak levels, voice can be given priority over data to ensure 
consistently  high  voice  quality  using  the  Differentiated  Services  (DiffServ)  Quality  of  Service  (QoS) 
protocol. This  is  an  end-to-end  requirement,  which  means  it  must  be  supported  at  various  points  on 
the network in order for the voice traffi c to receive the proper priority from every device it encounters.  
Another  way  to  enforce  Quality  of  Service  is  to  use  the  Resource  Reservation  Setup  Protocol  (RSVP).  
RSVP-enabled routers set aside bandwidth along the route from source to destination based on the IP 
addresses associated to the MultiVOIP gateways.
When  running  Voice  over  IP  on  the  public 
Internet,  the  issue  of  latency  cannot  be 
controlled  due  to  the  ever-changing  path  and 
router  hops  that  your  voice  packet  may  take 
before  it  reaches  its  destination.    However, 
the  MultiVOIP  gateway  does  a  good  job  of  not 
adding any additional latency through the box 
itself. Therefore,  if  you  have  a  good  Internet  service  provider,  and  they  are  able  to  provide  you  with  a 
quality of service guarantee, you should be able to manage any latency you may encounter.
If  you  have  concerns  about  latency  on  your  network,  or  the  public  Internet,  use  the  above  threshold 
chart to determine its possible affect on your voice quality.
Jitter  is  defi ned  as  the  variability  in  packet  arrival  at  the  destination.  Voice  packets  must  compete 
with  non  real-time  data  traffi c,  therefore,  if  there  are  bursts  of  traffi c  on  the  network,  they  can  result 
in  varied  arrival  times.  When  consecutive  voice  packets  arrive  at  irregular  intervals,  the  result  is  a 
distortion in the sound, which if severe, can make the speaker unintelligible.
The  MultiVOIP  gateway  utilizes  a  Dynamic  Jitter  Buffer  to  collect  voice  packets  from  the  IP  network, 
store  them,  and  shift  them  to  the  voice  processor  in  evenly  spaced  intervals.  During  high  latency 
periods,  the  jitter  buffer  size  is  dynamically  increased  to  receive  delayed  voice  packets.  During  low 
latency periods, the jitter buffer is dynamically decreased to minimize the end-to-end voice delay. 
Packet loss is the percentage of undelivered packets in the data network. When data packets are lost, a 
receiving computer can simply request a retransmission. When voice packets are lost, or arrive too late, 
they are discarded instead of retransmitted. The result is disconcerning gaps in the conversation (like a 
poor cell phone conversation).
The  MultiVOIP  gateway  utilizes  Forward  Error  Correction  to  increase  voice  quality  by  recovering  lost 
or  corrupted  packets.  The  current  Forward  Error  Correction  implementation  can  recover  one  of  two 
consecutive  lost/corrupted  packets  or  every  other  lost/corrupted  packet,  thereby  eliminating  any 
noticeable voice degradation.
Optimum Latency Thresholds and Voice Quality:
  Up to 150 ms = excellent
  150 - 250 ms = good
  250 - 350 ms = usually acceptable
    > 350 ms = depends on application 
						

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MultiVOIP  also  utilizes  Bad  Frame  Interpolation  to  increase  voice  quality  by  making  the  voice 
transmission more robust in bursty error environments. It interpolates lost/corrupted packets by using 
the previously received voice frames. Interpolation of one or two voice packets will not cause a noticeable 
degradation  in  voice  quality. Typically,  Bad  Frame  Interpolation  is  invoked  if  Forward  Error  Correction 
cannot  recover  the  lost/corrupted  packets.  By  utilizing  both  Forward  Error  Correction  and  Bad  Frame 
Interpolation, MultiVOIP is continually optimizing voice quality regardless of the conditions.
Security
“Will adding Voice over IP affect the security of my existing data network?”
On  a  private  network,  security  is  not  an  issue  because  the  network  is  private  to  outside  intruders.  If 
the VOIP  connection  is  over  the  Internet,  or  through  a VPN  connection,  the  network  security  will  not 
change. MultiVOIP does not interfere or change the way the current data security is set up. 
Standards
“Will the MultiVOIP gateway talk to other VOIP solutions?”
At  the  application  level,  standards  for  Voice  over  IP  interoperability  are  still  evolving.    The  H.323 
standard is the one most widely deployed and is the only approved protocol adopted by the International 
Telecommunications Union (ITU).  It is an umbrella standard that specifi es the components, protocols 
and procedures providing multimedia communication over packet-based networks.
Another  emerging  standard,  developed  by  the  Internet  Engineering  Task  Force  (IETF),  is  the  Session 
Initiation Protocol (SIP).  This protocol, designed specifi cally for VOIP applications, is gaining popularity 
in the area of IP phones and soft phones (MS Messenger) because of its simplicity.  
MultiVOIP  utilizes  both  the  H.323  and  SIP  protocols  to  provide  complete  interoperability  with  other 
Internet  telephony  solutions.    The  inbound  IP  call  protocol  is  automatically  detected  and  the  voice 
channel is dynamically confi gured to match.  The outbound IP call protocol is confi gured with the phone 
number allowing you the fl exibility to call H.323 or SIP devices from the same port.
Reliability
“Can you assure me MultiVOIP is going to work all of the time?”
Telecommunications managers have been accustomed to delivering a 99.999% reliable service. Because 
the  MultiVOIP  gateway  works  with  the  existing  phone  system,  there  is  little  risk  in  deployment.  If 
the  data  network  should  go  down,  or  if  all  the  VOIP  channels  were  busy,  the  user  can  always  revert 
automatically or manually to the standard PSTN to make the call.
“What happens if my LAN/WAN goes down?”
MultiVOIP  utilizes  a  feature  called  PSTN  fail-over  that  allows  it  to  automatically  route  calls  over  the 
PSTN network when the IP network is congested or completely down. This feature heightens reliability 
and  augments  QoS  when  conditions  threaten  to  undermine  voice  quality.  Utilizing  user  defi nable 
controls,  MultiVOIP  continually  checks  if  the  LAN/WAN  is  threatened  by  packet  loss,  jitter  or  latency, 
or to see if the network is completely down. If it detects a problem, MultiVOIP switches to “survivability 
mode”  transparently  routing  all  calls  over  PSTN  lines  connected  to  the  MultiVOIP  gateway.  MultiVOIP 
continues  to  monitor  the  connection  and  automatically  switches  back  to  the  LAN/WAN  once  the 
conditions improve. 
						

							Copyright © 2003 Multi-Tech Systems, Inc.  All rights reserved.11
Ease of Use
“Will my users require extensive training to use the MultiVOIP system?”
No, placing calls with MultiVOIP is like using your existing phone system. It uses single-stage dialing by 
utilizing a Uniform Dialing Plan that is consistent with the E.164 (PSTN) standard numbering plan. This 
includes  automatic  appending  and  stripping  of  digits  to  dialed  numbers  to  ensure  that  users  will  not 
require additional training to make VOIP calls.
Networking Dissimilar Proprietary PBX Systems
“Will MultiVOIP work when networking dissimilar proprietary PBX phone systems?”
Yes, as long as the PBX has analog extension ports, CO ports, E&M ports, or T1/E1/PRI cards available, or 
the ability to add cards with the appropriate interface. There is nothing proprietary about an analog or 
digital interface.  This is the benefi t of utilizing the MultiVOIP gateway. It simply bridges the two systems 
together. 
Supplementary Services
“Does the MultiVOIP support PBX-like features such as call transfer, call forwarding and call hold?”
Yes, MultiVOIP supports H.450 supplementary services to provide for call transfer, call forwarding, call 
hold, call waiting, and name identifi cation.  It also supports Q.SIG, an inter-PBX signaling protocol, for 
networking PBX supplementary services in a multi- or uni-vendor environment. In addition, MultiVOIP 
supports SIP extensions providing call forward and call transfer capabilities.
Management
“Can I manage my MultiVOIP gateways from a central location?”
Yes,  the  MultiVOIP  gateway  is  easily  managed  locally  using  a  windows-based  software  application 
or  remotely  by  the  central  offi ce  with  a  web  browser  or  SNMP.    Multi-Tech  also  includes  its  own 
SNMP  management  software  called  MultiVOIPManager,  which  provides  central  site  confi guration, 
management  and  call  monitoring  for  all  MultiVOIP  gateways  on  the  network.  It  utilizes  a  Windows 
interface that makes it easy to view events like usage tracking, live use reporting, call history, and voice 
quality statistics. In addition, MultiVOIPManager eases administration by automatically e-mailing call 
logs based on volume or time.
Plugging into the Voice and Data Network
“How does MultiVOIP plug into my existing voice and data network?”
For  maximum  investment  protection,  the  MultiVOIP  2-,  4-  and  8-port  models  accommodate  changing 
communication  needs  by  providing  a  programmable  FXS/FXO  and  E&M  interface  for  each  port.  This 
means you don’t have to worry about ordering the right interface to connect directly to the customer’s 
phones, fax machines, key phone systems or PBX system. On the digital MultiVOIP model, an industry 
standard  RJ-45  jack  is  provided  to 
connect  directly  to  either  a  digital 
port  on  the  PBX  or  directly  to  a T1/E1 
or  PRI  line.  On  the  data  network  side, 
the  MultiVOIP  gateway  simply  plugs 
into the Ethernet network.
PBX trunkPhone/Fax or PBX extensions 
						

							Copyright © 2003 Multi-Tech Systems, Inc.  All rights reserved.12
Port Confi guration
“How do I determine the number of ports I need and which MultiVOIP to order?”
You do not need a port for every telephone on the PBX system. You simply need to determine the calling 
ratio to determine how many ports you need at each location. The following guidelines can help you:
1)  If  you  are  replacing  tie/trunk  lines,  for  every  line  that  you  support  you  need  one  port  on  the 
MultiVOIP.
Ex.  4 Tie Lines = 4-port MultiVOIP (MVP410)
2)  If  you  are  not  using  tie/trunk  lines,  you  can  determine  the  ratio  based  on  the  location’s  long 
distance  communication  bills.  First,  determine  what  percentage  of  the  bill  is  used  for  intra-
offi ce  communication  (typically  between  25%  to  40%),  then  multiply  the  percentage  by  the 
number of available PSTN lines at the location. The result will determine the minimum number 
of ports needed.
Ex. Minneapolis, Corporate: 
  25% is intra-offi ce calling.  They have 16 lines.
  25% x 16 = 4
  Recommendation:  4-port MultiVOIP (MVP410)
Los Angeles Branch offi ce: 
  30% is intra-offi ce calling. They have 5 lines.
  30% x 5 = 1.5 
  Recommendation:  2-port MultiVOIP (MVP210)
London Branch offi ce:  
40% is intra-offi ce calling. They have 4 lines.
40% x 4 = 1.6 
Recommendation:  2-port MultiVOIP (MVP210)
If you do not know what percentage of the phone bill is being used for intra-offi ce communication, 
the rule of thumb is 30%.
If you need more than 16 ports, we recommend the digital MultiVOIP (MVP2410 or MVP3010).
For  a  worksheet  designed  to  help  you  calculate  your  customer’s  bandwidth  and  port  confi guration 
needs, reference our Confi guration Guide located in the back of this primer.
Gatekeeper Models
“When do I use a MultiVOIP Gateway/Gatekeeper (-G Model)?”
The  MultiVOIP  Gateway/Gatekeeper  (-G  Model)  includes  an  integrated  gatekeeper  to  facilitate  call 
management  in  a Voice  over  IP  network. These  cost-effective  MultiVOIP  gateways  provide  centralized 
phone  book  management  as  well  as  deliver  the  power  to  defi ne  and  control  how  H.323  voice  traffi c  is 
managed  over  IP  networks. With  the  integrated  gatekeeper,  network  managers  can  confi gure,  monitor 
and manage the activity of registered end points. In addition, they can set policies and control network 
resources, such as bandwidth usage, to ensure optimal implementation.
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							Copyright © 2003 Multi-Tech Systems, Inc.  All rights reserved.13
Voice over IP Applications
MultiVOIP  is  ideal  for  multi-location  businesses  looking  to  reduce  toll  charges  associated  with  intra-
offi ce  calling.  It  is  designed  to  help  a  company  maximize  investments  they’ve  already  made  in  their 
data network infrastructure and voice equipment. Using our company example, the following are some  
of the many applications for a VOIP network:
Offi ce-to-Offi ce Communication
A VOIP network can be as small as two offi ces or as large as hundreds of offi ces. Each offi ce installs 
and  confi gures  a  VOIP  solution  on  their  network  to  begin  placing  calls  or  sending  faxes  to  other 
offi ces  on  the  VOIP  network.  This  allows  a  company  to  extend  its  telecommunications  network  to 
remote  offi ces  without  the  expense  of  replacing  the  phone  system  at  each  location.  Our  company 
example shows a typical three offi ce VOIP network.
Create Off-Premise Extensions for Telecommuters
Extend the reach of a customer’s PBX into home offi ce locations. Simply connect a VOIP solution to 
the PBX at the corporate offi ce, and another VOIP solution at the home offi ce. Now, anyone can place 
calls to the home offi ce by dialing an extension number. And, the home offi ce can dial others on the 
VOIP network without incurring long distance charges. In our company example, we have now added 
two telecommuters to the VOIP network.
Offi ce-to-Offi ce
Off-Premise Extensions 
						

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Wireless Connections
To  extend  a  customer’s  PBX  to  a  building  across  the  street,  utilize  a  wireless  bridge  to  connect  the 
two networks. Now, your customer has voice and data connectivity without laying cables or paying 
monthly  charges  for  dedicated  lines.  Building  off  our  example VOIP  network,  we  have  now  added  a 
wireless connection to the company’s warehouse.
Wireless Building-to-Building 
						

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Confi guring a MultiVOIP Network
Now that you have a basic understanding of Voice over IP, the MultiVOIP gateway, and its applications, 
the  next  step  is  to  learn  how  easy  it  is  to  confi gure  the  solution  around  existing  telephone  and  data 
networks.
Confi guring the Telephony Interface
Let’s fi rst discuss how to confi gure the MultiVOIP gateway to the various telephony options. We will look 
at both the analog and digital MultiVOIP solution.
Confi guring an Analog MultiVOIP:
The MultiVOIP gateway is equipped to support one of three voice-port signaling types*:
1.  FXS  (foreign  exchange  station)  interface:  connects  directly  to  phones,  faxes,  and  CO  ports  on 
PBXs or key telephone systems (KTS)
2. FXO (foreign exchange offi ce) interface: connects directly to an analog PBX extension, PSTN, or 
KTS extensions
3. E&M interface (Ear and Mouth): connects directly to analog PBX trunk ports
*  MVP130 supports FXS and FXO only.
The type of phone equipment that you use to connect to the MultiVOIP gateway will determine which 
interface port you will use:
You will note that with a key telephone system and a PBX, you have a couple of interface options.  The 
interface  that  you  use  will  create  a  different  path  and  dialing  procedures  for  the Voice  over  IP  call. 
In general, MultiVOIP does not modify the behavior of the telephone equipment. It simply provides 
a connection to the IP data network instead of the PSTN and passes along the equipment’s features 
and functionality. 
						

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Key Telephone System and PBX Interface Options:
1.FXS interface - Use this interface when connecting the line side (CO port) of your key telephone 
system or PBX to the MultiVOIP gateway. The system will act similarly to connecting directly to 
a CO line. Here’s how it works:
Outgoing calls: The caller needs to access the MultiVOIP gateway in order to receive a dial tone.  
An  easy  approach  would  be  to  associate  a  button  on  the  phone  system  that  will  access  the 
MultiVOIP  gateway. At  this  point,  MultiVOIP  will  generate  a  dial  tone  and  the  caller  can  call 
any other location on the network that is also equipped with a MultiVOIP gateway.
Incoming  calls:  MultiVOIP  will  generate  the  ring  signal,  just  like  an  outside  CO  line.  At  this 
point, you can use the features of the telephone system to determine how to handle the call. 
For  example,  on  a  key  telephone  system  all  of  the  phones  may  be  programmed  to  light  up. 
On a more advanced phone system, you can route the incoming call to a receptionist or to an 
Interactive Voice Response (IVR) system.
2.FXO  interface  -  Use  this  interface  when  connecting  the  station  side  (extension  port)  of  your 
telephone system to the MultiVOIP gateway.
Outgoing calls: The caller will need to access the extension in order to receive a dial tone from 
the  MultiVOIP  gateway.  This  can  be  done  by  simply  dialing  the  extension  number  that  the 
MultiVOIP  is  connected  to,  or  by  preprogramming  a  button  to  access  the  extension.  At  this 
point,  the  MultiVOIP  gateway  will  generate  a  second  dial  tone.  Now,  the  caller  can  call  any 
other location on the network that is also equipped with a MultiVOIP. 
Incoming  calls: The  MultiVOIP  gateway  will  utilize  an  extension  on  the  phone  system. At  this 
point, the caller will hear an extension dial tone generated by the remote phone system. The 
caller can then complete the call by dialing any other extension on the phone system. An easy 
approach to eliminate the need for this “two stage” dialing would be to use the phone system’s 
ability to route the incoming call immediately to a reception desk or IVR system.
3.E&M interface - E&M is the preferred interface, on a PBX system, because it provides the most 
reliable and quickest disconnect among calls allowing the MultiVOIP to re-establish the port for 
availability. It is also the interface used when you are using the MultiVOIP solution to replace tie 
lines. For outgoing and incoming calls, the dialing behavior is the same as the FXO interface (as 
described above). 
All channels on the MultiVOIP gateway do not have to be confi gured the same way. For example, one 
channel could be connected to an extension line off the PBX (using the FXO interface), and the other 
channel connected directly to a fax machine (using the FXS interface).
Digital MultiVOIP Confi guration:
The  digital  MultiVOIP  uses  a  T1,  E1  or  PRI  interface  to  connect  to  a  PBX  system.  For  outgoing  and 
incoming calls, the dialing behavior is the same as the FXO interface (as described above). Multi-Tech 
recommends utilizing a digital MultiVOIP solution when you are connecting 16 or more lines. 
						

							Copyright © 2003 Multi-Tech Systems, Inc.  All rights reserved.17
Building the VOIP Dialing Plan
MultiVOIP provides single stage dialing by utilizing a Uniform Dialing Plan that is consistent with both 
the  PBX  and  the  E.164  (PSTN)  standard  numbering  plan.    This  means  placing  calls  is  like  using  your 
existing phone system, no user training is needed.
In order to accomplish this, MultiVOIP utilizes an Inbound and Outbound phonebook.  In our company 
example,  the  system  allows  employees  in  any  offi ce  to  dial  employees  in  any  other  offi ce  using  only 
three digits.  The following is a worksheet used to layout this company’s dialing plan.
A  more  sophisticated  phonebook  setup  can  be  created  that  requires  an  “intelligent”  PBX  and  the 
MultiVOIP gateway to automatically append and strip digits.  So, for example, when an employee dials 
another  remote  company  location  using  a  standard  12-digit  number,  the  PBX  would  know  to  hand 
that call to the MultiVOIP gateway.  MultiVOIP, in turn, would strip the unnecessary 5-digit destination 
pattern (9+1+area code) and direct the call to the IP address of the remote MultiVOIP.  For details on this 
type of dialing plan, refer to the user’s guide. 
						

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When  confi guring  the  MultiVOIP  gateway 
for  the  network,  you  must  identify,  within 
the  software,  its  IP  address,  subnet  mask, 
and gateway address. The IP address is your 
unique LAN IP address. The subnet mask is 
the number that identifi es the sub network 
to  which  your  MultiVOIP  is  connected.  The 
gateway  address  is  the  IP  address  of  the 
device  connecting  your  MultiVOIP  to  the 
Internet/intranet.  Each  MultiVOIP  on  the 
network  will  need  to  be  confi gured  this 
way  so  that  the  phone  directory  can  be 
mapped  to  the  IP  address  of  the  individual 
MultiVOIPs on the network.
Deploying the VOIP Network
The VOIP administrator can take each individual MultiVOIP and pre-confi gure it before sending it to the 
remote sites. The remote site administrators need only connect power to the pre-confi gured MultiVOIPs, 
and  connect  them  to  the  Ethernet  LAN  and  predefi ned  telephone  equipment.  At  this  time,  the  VOIP 
network will be fully operational. 
Advanced Feature Confi guration
The  MultiVOIP  software  provides  a  number  of  advanced  features  that  can  be  confi gured  to  enhance 
voice quality. Our recommendation is to use the factory defaults. Most users fi nd these are more than 
adequate for the application.