Digium’s new D40, D50 and D70 IP Desk Phones are geared towards utilizing Switchvox (and specifically Asterisk) features and capabilities. The system will be able to find and configure these phones upon deployment saving time and energy on any future installations. One of the biggest concerns end-users had when faced with the decision to purchase an Asterisk system was the connotation that products were pieced together to make a solution as compared to the Avaya/Shortel/Cisco solutions that customers are offered. Now with a competitively priced Unified Communications platform standardized on the Digium Brand for the PBX and endpoints, a whole new breed of Asterisk dealers will emerge to have a true chance to make an impact on the SMB telephony market.

Digium is announcing these devices at this year’s IT EXPO East in Miami and ABP is the preferred Distribution Partner for the complete Digium solution with Switchvox and the new IP Phones. The current lineup is the entry level D40, the mid level D50 and the receptionist/executive level D70. The D40 comes with 2 line keys, 4 XML programmable keys and a 240×120 pixel screen. There are navigation buttons as well as dedicated keys to normal call features such as transfer, hold, redial, messaging plus a volume bar located at the bottom of the phone. The D50 has 4 line keys and 4 XML programmable keys and a 240×120 pixel screen as well. This model adds 10 BLF keys with designation strips as well as additional buttons dedicated for conference and intercom functions. Lastly, the D70 Executive level IP Phone comes with 6 line keys, 4 XML programmable buttons, 10×10 BLF keys with LCD display plus an optional sidecar module. It boasts a 320×160 pixel screen and adds on new buttons for Do Not Distrub, Applications, directory and menu. All 3 models of phones come with Backlit Displays and they all support HD Voice. The combination of these 3 Digium IP Phones on Switchvox creates a powerful selling solution to meet the needs of any size potential customer.

In the new firmware of Switchvox, there is a menu option when creating new extensions specifically for Digium IP Phones. During the training session at ABP, Digium showed how quick and easy it was to do the first time configuration of  new these new devices and to tie them to a specific user or extension. The Switchvox system was already one of the most powerful IP Phone Systems on the market for SMB customers but now ABP and Digium are certifying VARs and Integrators to become true Digium Dealers. We are currently implementing online training for our customers to help get anyone interested up to speed to hopefully become a Digium Select Partner one day. Please contact sales@abptech.com with any inquiries on pricing or building solutions with Switchvox and these Great new IP Phones from Digium.

(Please click below image to enlarge)

New Menu Option in Switchvox for Digium IP Phones

The

Yes UC is nothing new and really, you may not have sold this type of voice solution before. However, things have changed. There is finally a rich enough feature set and experience level to roll out Lync-based phone solutions with true standalone IP phones available as desktop endpoints along with headset based client solutions.  Customers now have the choice to build a much more conventional feeling solution with all the advantages of a Unified Communications Platform that is heavily computer centric. This allows for many choices and provides for the richest spread in features, handset devices and overall potential. Every single Microsoft Partner will be able to sell a competitive phone platform now based around the new selection of IP Phone devices.

ABP is the choice Distribution Partner for any Lync customers today and tomorrow. We can immediately offer Microsoft partners multiple brands and products to suit the need of any customer looking for a supported device. We work with top of the line and certified vendors such as Aastra, AudioCodes, ClearOne, Jabra, NET, Patton, Plantronics, Polycom, snom (with dual SIP/Lync registration capabilities) and the unique StartReady appliances that host Lync. As a one stop solution for all Lync endpoints and basic support, we can help Gold Partners lock in their business now. Don’t lose these high stakes clients by limiting yourself. ABP will help you provide whatever chosen Lync device needed as well as offer you the top of the line support you deserve. Make sure you contact your representative at ABP for test equipment and to discuss any Lync applications because it’s not a question of if this platform will take off, but when. Will you be prepared?

More than 100,000 visitors also came to see where the technology is going. A lot of innovation was displayed everywhere. Since my focus is in networking, I was more interested in the “next new networking thing”. Based on our solid distribution partnership, Draytek invited ABPTech to be part of Draytek’s booth. In addition to my exhibitor role, I had the opportunity to walk the halls as visitor.

Router manufacturers have been constantly innovating and bringing new products to market. New state of the art CPE devices and routers technology have improved enormously in recent years. With the advance of processing power we now see routers that are much more intelligent than their counterparts from last decade. Just like most electronic products, routers are now smaller, more intelligent, more cost efficient and environment friendlier.

The next generation routers brought by Draytek Corp shinned. 4G, Gigabit, fiber and Vdsl interfaces are part of many models. The new breed of routers are designed to better work and allow us to safely take advantage of emergent technologies such as Cloud Based Services, IPV6 and 4G LTE networks.

Routers must have redundant WAN ports for load balancing and failover. Amazing new cloud based applications won’t work if the internet it down. Today’s routers must be able to police the bandwidth and kind of traffic or applications running on it. Bandwidth management, true QOS and Web Content Filtering are part of next generation routers to make sure our resources, including employees’ time and network speed, are utilized in benefit of the company.

In addition to the above, other features such as VPN, Firewall, Multi LAN, VLan and WiFi are incorporated into the same box reducing dramatically the energy consumption and clutter in the telecom rack or server room. For multisite deployments a cloud based, centralized management and monitoring interface is desirable.

Draytek 2920n

ABPTech’s upcoming webinar will focus on Draytek’s next generation routers and advanced network features. Join us for our Jan. 26th @ 11:00 AM webinar. We’ll extend a special offer to the webinar attendees.

Henry Castillo

This document applies to Patton FXS gateways configured to hand-off SIP trunks to legacy analog PBXs or individual analog phones.
In this scenario we have a four port FXS SN4114 unit in which the incoming calls from Broadvox SIP trunk will be routed to a specific FXS port based on the DID. In this example DID1 will go to first port; DID2 will go to second port; DiD3 will go to third port and DID4 will go to fourth port.

Copy the configuration file provided below to notepad and make the replacements as necessary, then upload that file to your Patton using the gateway’s web interface.

Following the same logic you can easily create the configuration file for an Patton with any number of FXS ports.
You need to have a Registration type of account from Broadvox , the BTN/telephone number, the list of DIDs and the password and server address as provided by Broadvox:

PHONE_NUMBER (BTN=DID1) and other DIDs (DID2, DID3, DID4)
PASSWORD
SERVER_IP_ADDRESS_OR_FQDN

#File start—————————————————-#
#                                                                                                          #
# SN4114/JS/EUI                                                                       #
# R5.8 2011-09-05 H323 SIP FXS FXO                             #
#—————————————————————-#

cli version 3.20
dns-client server DNS_SERVER_Address (i.e. 8.8.8.8)
dns-relay
webserver port 80 language en

system

ic voice 0
low-bitrate-codec g729

profile napt NAPT

profile ppp default

profile call-progress-tone US_Dialtone
play 1 1000 350 -13 440 -13

profile call-progress-tone US_Alertingtone
play 1 1000 440 -19 480 -19
pause 2 3000

profile call-progress-tone US_Busytone
play 1 500 480 -24 620 -24
pause 2 500

profile call-progress-tone US_Releasetone
play 1 250 480 -24 620 -24
pause 2 250

profile tone-set default
profile tone-set US
map call-progress-tone dial-tone US_Dialtone
map call-progress-tone ringback-tone US_Alertingtone
map call-progress-tone busy-tone US_Busytone
map call-progress-tone release-tone US_Releasetone
map call-progress-tone congestion-tone US_Busytone

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
codec 3 g729 rx-length 30 tx-length 30
fax transmission 1 relay t38-udp

profile pstn default

profile ringing-cadence default
play 1 1000
pause 2 4000

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_LAN
ipaddress IP_ADDRESS SUBNET_MASK (i.e. 192.168.0.50 255.255.255.0 )
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
route 0.0.0.0 0.0.0.0 DEFAULT_GATEWAY (i.e 192.168.0.254) 0

context cs switch
digit-collection timeout 3
address-completion timeout 3

routing-table called-e164 IN
route DID1 (i.e. 4795555555) dest-interface IF_FXS_0
route DID2 (i.e.4795555556) dest-interface IF_FXS_1
route DID3 (i.e.4795555557) dest-interface IF_FXS_2
route DID4 (i.e.4795555558) dest-interface IF_FXS_3

routing-table called-e164 FXS_to_SIP
route .T dest-interface IF_SIP_0

interface sip IF_SIP_0
bind context sip-gateway GW_SIP_0
route call dest-table IN
remote SERVER_IP_ADDRESS_OR_FQDN (i.e. ld01-03.fs.broadvox.net)
early-connect
early-disconnect
address-translation incoming-call called-e164 to-header

interface fxs IF_FXS_0
route call dest-table FXS_to_SIP
subscriber-number DID1 (i.e. 4795555555)
use profile tone-set US

interface fxs IF_FXS_1
route call dest-table FXS_to_SIP
subscriber-number DID2 (i.e. 4795555556)
use profile tone-set US

interface fxs IF_FXS_2
route call dest-table FXS_to_SIP
subscriber-number DID3 (i.e 4795555557 )
use profile tone-set US

interface fxs IF_FXS_3
route call dest-table FXS_to_SIP
subscriber-number DID4 (i.e 4795555558 )
use profile tone-set US

service hunt-group hunt_in
cyclic
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
route call 1 dest-interface IF_FXS_0
route call 2 dest-interface IF_FXS_1
route call 3 dest-interface IF_FXS_2
route call 4 dest-interface IF_FXS_3

context cs switch
no shutdown

authentication-service SOFTSWITCH_EXT
username DID1 (i.e 4795555555 ) password PASSWORD (i.e 3KOas9eR5)

location-service LS_SN
domain 1 SERVER_IP_ADDRESS_OR_FQDN (i.e. ld01-03.fs.broadvox.net)

identity-group IDENT-GROUP

authentication outbound
authenticate 1 authentication-service SOFTSWITCH_EXT

registration outbound
registrar SERVER_IP_ADDRESS_OR_FQDN (i.e.ld01-03.fs.broadvox.net)
lifetime 3600
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

identity DID1 (i.e 4795555555 ) inherits IDENT-GROUP

registration outbound

context sip-gateway GW_SIP_0

interface LAN
bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_0
bind location-service LS_SN
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port fxs 0 0
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_0 switch
no shutdown

port fxs 0 1
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_1 switch
no shutdown

port fxs 0 2
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_2 switch
no shutdown

port fxs 0 3
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_3 switch
no shutdown

# File end #

 
If you have problems or have a custom scenario please contact ABP Technology support.

This document applies to Patton FXS gateways configured to hand-off SIP trunks to legacy analog PBXs.

In this scenario we have a two port FXS SN4112 unit in which all the incoming calls from Broadvox SIP trunk will be routed to the first available FXS port. If you want to configure a 4 port Patton gateway just uncomment the definitions for FXS ports 3 and 4. Following the same logic you can easily create the configuration file for an Patton with any number of FXS ports.

You need to have a Registration type of account from Broadvox , the BTN/telephone number, the password and server address as provided by Broadvox:

PHONE_NUMBER
PASSWORD
PROVIDER_SERVER_IP_ADDRESS_OR_FQDN

Copy the configuration below to a text file, modify highlighted red text, and import to the Patton unit using the Web GUI.

#file start#
#----------------------------------------------------------------#                                            #
# configuration file                                             #
# Modify before loading                                         #

#----------------------------------------------------------------#

cli version 3.20
dns-client server DNS_SERVER_Address   (i.e. 8.8.8.8)
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4

system

  ic voice 0
    low-bitrate-codec g729

profile napt NAPT

profile ppp default

profile call-progress-tone US_Dialtone
  play 1 1000 350 -13 440 -13

profile call-progress-tone US_Alertingtone
  play 1 1000 440 -19 480 -19
  pause 2 3000

profile call-progress-tone US_Busytone
  play 1 500 480 -24 620 -24
  pause 2 500

profile call-progress-tone US_Releasetone
  play 1 250 480 -24 620 -24
  pause 2 250

profile tone-set default
profile tone-set US
  map call-progress-tone dial-tone US_Dialtone
  map call-progress-tone ringback-tone US_Alertingtone
  map call-progress-tone busy-tone US_Busytone
  map call-progress-tone release-tone US_Releasetone
  map call-progress-tone congestion-tone US_Busytone

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  codec 3 g729 rx-length 30 tx-length 30
  fax transmission 1 relay t38-udp

profile pstn default

profile ringing-cadence default
  play 1 1000
  pause 2 4000

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_LAN
    ipaddress IP_ADDRESS SUBNET_MASK (i.e. 192.168.0.50 255.255.255.0 )
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context ip router
  route 0.0.0.0 0.0.0.0 DEFAULT_GATEWAY (i.e 192.168.0.254) 0
# notice the "0" at the end of previous line, it is required
context cs switch
  digit-collection timeout 3
  address-completion timeout 3

  routing-table called-e164 IN
    route .T dest-service hunt_in

  routing-table called-e164 FXS_to_SIP
    route .T dest-interface IF_SIP_0

  interface sip IF_SIP_0
    bind context sip-gateway GW_SIP_0
    route call dest-table IN
    remote PROVIDER_SERVER_IP_ADDRESS_OR_FQDN  (i.e. 208.93.224.229)
    early-connect
    early-disconnect

  interface fxs IF_FXS_0
    route call dest-table FXS_to_SIP
    subscriber-number PHONE_NUMBER (i.e 4795555555 )
    use profile tone-set US

  interface fxs IF_FXS_1
    route call dest-table FXS_to_SIP
    subscriber-number PHONE_NUMBER (i.e 4795555555 )
    use profile tone-set US
#if your gateways has more ports add them here

#interface fxs IF_FXS_2
#    route call dest-table FXS_to_SIP
#    subscriber-number PHONE_NUMBER (i.e 4795555555)
#    use profile tone-set US

#interface fxs IF_FXS_3
#    route call dest-table FXS_to_SIP
#    subscriber-number PHONE_NUMBER (i.e 4795555555 )
#    use profile tone-set US

  service hunt-group hunt_in
    cyclic
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_FXS_0
    route call 2 dest-interface IF_FXS_1
#if your gateways has more ports add them here
#   route call 3 dest-interface IF_FXS_2
#   route call 4 dest-interface IF_FXS_3
  context cs switch
  no shutdown

authentication-service SOFTSWITCH_EXT
  username PHONE_NUMBER (i.e 4795554520 ) password PASSWORD ( i.e 2kz4Ed5Ea1)

location-service LS_SN
  domain 1 PROVIDER_SERVER_IP_ADDRESS_OR_FQDN  (i.e. 208.93.224.229)
  identity-group IDENT-GROUP

    authentication outbound
      authenticate 1 authentication-service SOFTSWITCH_EXT

    registration outbound
      registrar PROVIDER_SERVER_IP_ADDRESS_OR_FQDN  (i.e. 208.93.224.229)
      register auto

  identity PHONE_NUMBER (i.e 4795554520 ) inherits IDENT-GROUP

    registration outbound

context sip-gateway GW_SIP_0

  interface LAN
    bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_0
  bind location-service LS_SN
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface IF_IP_LAN router
  no shutdown

port fxs 0 0
  use profile fxs us
  encapsulation cc-fxs
  bind interface IF_FXS_0 switch
  no shutdown

port fxs 0 1
  use profile fxs us
  encapsulation cc-fxs
  bind interface IF_FXS_1 switch
  no shutdown

#if your gateways has more ports add them here
#
#port fxs 0 2
#  use profile fxs us
#  encapsulation cc-fxs
#  bind interface IF_FXS_1 switch
#  no shutdown

#port fxs 0 3
#  use profile fxs us
#  encapsulation cc-fxs
#  bind interface IF_FXS_1 switch
#  no shutdown

#file end#

If you have problems or have a more custom scenario please contact ABP Technology support.

Henry

The TE820 is an eight-port digital interface card. It can handle up to 8 T1, E1 or J1 connections. That’s up to 192 concurrent calls in T1 or J1 mode, or 240 calls in E1 mode, making it the highest-density card made for use with Asterisk. As an optional enhancement to TE820, we are also launching the VPMOCT256, an echo cancellation module that can handle up to 256 concurrent calls. The TE820 will be available in PCI Express form factor.
The TE800 Series cards consists of 4 RJ45 ports and eight status LEDs. The ports are used for connecting T1, E1, or J1 cables via Y-adapter dongles, which are included in the shipment of the TE800 Series card. The TE800 Series cards can be used without Y-adapter dongles, but only spans 1 through 4 will be accessible.

Digium’s Redundancy Series (aka “R-Series”) products are a family of inexpensive, versatile appliances that enable physical-layer failover of analog (POTS) and digital (T1, E1, PRI or BRI) telephony connections to Asterisk-based communications systems. The R-Series devices allows Asterisk administrators to create high-availability solutions that insure uninterrupted business communications, even in the event of a catastrophic hardware or software failure.

Patton’s award-winning Ethernet Extender solutions extend long-range Ethernet farther & faster.

Use CopperLink™ Ethernet extension solutions to extend Ethernet services to your users located far beyond distance limitations. With Patton’s Ethernet Extender solutions you can extend long-range Ethernet over existing voice-grade copper wiring, up to 5 miles over a single twisted pair. Contact sales@abptech.com or call 972-831-1600 to order your Patton Ethernet Extenders for your next project.

CopperLink™ Ruggedized Ethernet Extenders

CopperLink-T™ T1 and E1 Network Extenders

Think twice when implementing VoIP and other Cloud Services, you don’t want to find yourself in this situation:

Do not put all eggs in one basket, install a dual Wan capable router.

Why put all Eggs in One Basket?  Run you Business  on a State of the Art Dual WAN Load Balancing Router enjoying twice the speed everyday and never be down!

Past experiences on Internet failing and bringing down email, CRM, ERP or other cloud based services, make Companies hesitate when deciding to move to a VoIP solution. Some companies would delay the decision to save money and move to VoIP as both approaches, hosted PBX or SIP trunking, rely 100% on Internet connectivity.

In addition to WAN failover, the Dual Wan router must support true load balancing. That way additional bandwidth from secondary WAN is aggregated and not just sitting idle.  The router must be sophisticated enough to recognize and remember sessions and perform session Load balancing. Stateless Packet load balancing would break Voip, https and other protocols.

QoS and packet prioritization is not enough, make sure your router can do Bandwidth Management.

There is different approaches to QoS. Class of service or Diffserv are many times called QoS. Bandwidth Management can also be confused with QoS. Bandwidth Management is what we need when dealing with VoIP.

Most networking devices claim to support QoS, even layer 2 switches. For some manufacturers QoS mean setting a flag in all packets, others would also classify and apply prioritization rules before forwarding the packets. QoS is no help when upstream modem or router start dropping packets when the link is flooded.

In any case only routers that support true Bandwidth Management can limit the bandwidth used by non-time sensitive applications and dynamically allocate enough to VoIP traffic. The router must be capable to slow down bandwidth hogs and connections that are less important.

Even better, what if the router could also help manage the content going in and out your network? Content Management capability in the router is a great advantage. That way, undesired traffic, the users can be generating, does not even start. Only expensive high-end networking appliances supported these features until now.  ABP this year introduced Draytek, a value alternative that beats the features of high-priced routers while offering more innovative features and with NO PER USER LICENCES.

Separate VoIP from data LAN for performance and security, use a router that supports multiple Vlans and Lan subnets.

Computers can get viruses and Trojans, users many times install software that can impacts the network such as P2P or streaming applications. There can be many things that can go wrong in the data network as the applications that run over it change frequently. A dedicated Voice VLAN  subnet keeps the Voice network stable and voice quality high regardless of congestion generated by applications such as  IP Surveillance, email, video streaming or other data Transmissions.

An elevated number of Broadcasts in the Ethernet network or hacking attempts can be isolated from Voice LAN with the use of VLANs. Most managed switches support Vlans but makes sense only if the router does too. The router should be able to handle multiple VLANs and IP ranges including multi-subnet DHCP server and layer 3 routing between subnets.

Draytek 2920 supports up to 4 independent LANs separated by physical ports or thru Vlans. That means you can eliminate equipment and have a secure solution with a much simpler setup and maintenance. Draytek can be configured so routing between LAN subnets can transparent or firewalled.

A well thought approach with a happy ending is now affordable:

Click HERE for other articles on Draytek

Henry Castillo

Technical Director @ ABPTECH

What are the different types of locks?

How should I connect the lock?

How do I configure the jumpers?

Non-12V or high power door strike

If you have a door strike that has a higher threshold than 12v the Helios has a built in relay switch. This circuit can be independent of the Helios allowing a high voltage/power source connection which will only power the door strike. The images below will help you set up the lock and configure the jumpers correctly.

Note: the configuration connector in the picture below is the location on the circuit board where you connect the jumpers.

Magnetic lock

If you have a magnetic lock that requires constant voltage in order to stay locked the Helios’s built in relay switch will work for this as well. This Method can also be used for a door strike that requires continuous power. The advantage of this type of door strike is if your buildings power goes out the door will unlock.

12v door strike

For a 12v door strike installation you can connect the door strike directly to the terminal block. This method has the advantage of being powered internally and does not require a second source of power.

Note: If your door strike is 12v and does not work when connected to the Helios does not mean your unit is broken. It is potential that your door strike requires more wattage than the unit can supply.  In this case you can fix this by connecting an external power supply (first method). Connecting a 12volt power supply with higher amperage will give you the needed power for the door strike. A POE switch meeting the 802.1af standard will only put out 15 watts of power maximum.

Watts calculation:

Volts X Amps = Watts

Users place high value on being able to watch the cameras from anywhere. Business owners can check their business operation or keep an eye on inventory from home or when travelling. Cameras are accessible even to handheld devices, there is a number of Apps that make really simple watch life video streams from mobile 3G or 4G devices. In larger multi-site deployments, Centralized remote video monitoring is becoming a much more efficient and cost effective approach.

Moreover, IP cameras and IP enabled surveillance systems have gotten so smart that they can send notifications on specific events in many different ways. i.e.  emails, sms, ftp uploads, and even phone calls. Video clips and snapshots  can also be conveniently moved offsite to headquarters or cloud based storage.

All the above benefits of new surveillance technologies have something in common: All depend 100% on IP networks and Wide Area Network access, in which the router used plays a very important rol.

Other elements such as Power and network switches are critical as well but that is matter of a different blog entry, in any case a good quality POE switch and power backup such us SMC and  Minuteman respectively allow to offer a solid infrastructure by just physically connecting the cables.

Resources usage in Converged Networks needs policing

“The Router” needs to be smart to handle converged networks that include data and voice traffic. Few Megapixel cameras can quickly saturate any available bandwidth rendering all other network services in that site down. Even when the camera can be configured for reduced frame-rate or resolution for remote viewing, too many camera viewers from the WAN can always cause problems.

“The Router” needs to be able to prioritize traffic and enforce QoS rules, and bandwidth limits called traffic shaping. Even better, dynamic rules that adjust based on schedule can allow for heavier video uploads during non-business hours. On the other hand we don’t need other applications taking over the available bandwidth completely.

Other common requirement in converged networks is the ability to virtually separate internal networks with the use of VLANs. Internal users and other applications traffic can be separated for increased performance and network security. A random internal user cannot access the subset where all the cameras and surveillance equipment is. Only advanced equipment provide multiple VLAN and IP subnets, and routing and firewall functions to control who can access what.

An Internet link fails but connectivity continues

“The Router” needs to support redundant Internet links specially when critical alarms need to be delivered. A secondary wireless ISP would prevent and intruder from cutting the wires to easily disconnect the cameras from the Internet and  disable all sophistication built in the surveillance system for remote alerting or video viewing. The router would detect when one link is down and re-route Ip traffic to the working WAN transparently for the cameras and other internal network devices.

Remote connections made easy and secure

“The Router” needs to allow easy remote connections to the LAN so the cameras can be reached from The Internet in a secure way. For a small number of cameras and when network security is not a real concern, opening ports in the router/firewall and port forwarding is easy to implement but the easiest and more secure is offering a VPN connection. Starting a secure VPN to the LAN puts us virtually  in the LAN with access to all local IPs without having to open ports. The router acts as a VPN server and should work with common VPN clients like the ones built in all Windows PCs, Apple MACs and even Iphones. Additional complexity to access a network remotely is added when the public IP addresses associated to the site is not static, to overcome that, Dynamic Dns is used to map a changing IP addresses on the router to a fixed domain name. “The Router” needs to have a built in DynDns client.

Temporary, mobile, isolated surveillance sites ? Use 4G

Connecting to the Internet thru 4G networks is many times the only option. Construction sites where is no demark point to get conventional ISPs or new buildings that can’t wait a month to get service can immediately be hooked to the Internet using a Router that connect to 4G internet providers such us Verizon or Clear. “The Router” has a USB port that can be used to connect 4G USB dongles/modem from providers like Verizon or Clear to allow sharing the 4G access among multiple devices in the LAN.

“The Router”

In addition to all the above, the router needs to be reliable and not expensive.

The only Router brand we found that complies with the above that has the smarts needed for a smart surveillance system is Draytek. More Information on Draytek can be found in this other blogs.

Henry Castillo

Technical Director @ ABPTech