A Digitally Enhanced Cordless Telephony (DECT) VoIP base station
paired with DECT handsets provide a secure and dependable wireless
communication solution. To the user community in any industry, the DECT
air interface/DECT radio technology via dedicated and reserved radio
frequency bands are less likely to suffer from radio frequency (RF)
interference. They provide mobility, high voice quality and security for
enterprise telephony applications. DECT VoIP based technology opens
many new avenues to support flexible and simple enterprise business
voice applications. From an end user perspective, benefits accrue when
both are combined on one platform, as it takes their communication and
collaboration capabilities to another level, even when they are on the
move.
This solution supports multiple SIP accounts and multiple lines per
handset. Each handset can map to any SIP account and each SIP account
can map to any handset. You can set up hunt group modes, such as
parallel mode, circular mode and linear mode. In parallel mode all the
phones ring concurrently. Once the call is answered on one, you can then
place calls from the others. In circular mode, all handsets ring
sequentially starting from the one that answered the last call. In
linear mode, all the handsets ring sequentially in a pre-designated
order. Push-to-talk (PTT), a two-way radio type service, enables simple
instantaneous and seamless voice communication with the push of a
button. We can now carry out upgrades, updates and change settings
wirelessly over the air.
Let us now look at products from Grandstream Networks, a provider of DECT VoIP solutions.
Up to 10 SIP accounts and 10 lines per handset, 3-way audio conferencing and push-to-talk button for instant voice communication
Codecs:G.722 codec for HD audio and G.726 codec for narrow band
audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via
companion base station), AEC, AGC, Ambient noise reduction on handset
mic, advanced noise suppression for incoming audio
500 hours of standby time and 40 hours of talk time
Up to 10 SIP accounts and 10 lines per handset, 3-way audio
conferencing and configurable push-to-talk button for instant voice
communication
Codecs:G.722 codec for HD audio and G.726 codec for narrow band
audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via
companion base station) AEC, AGC, Ambient noise reduction on handset
mic, advanced noise suppression for incoming audio
250 hours of standby time and 20 hours of talk time
Let us now conduct functional and interoperability testing of the
DP752 base station along with the DP730 and DP722 wireless IP phones.
FUNCTIONAL TESTING
The test setup will consist of the Grandstream Networks DP752 DECT
VoIP Base Station with the DECT wireless phones DP 730, DP 722 running
software release version: Version 1.0.9.1 and the following devices and
systems (Please refer to Fig. B):
Public SIP trunk services;
Patton SmartNode SN5570 Series eSBC (Enterprise Session Border
Controller) running software version Trinity 3.15.1-19021 will provide
SIP registrar services and SIP demarcation from the public SIP trunk
provider while maintaining interoperability and interconnectivity;
Microsemi PDS-208G Digital Ceiling PoE switch with software release
version 2.53 for efficient network connectivity and PoE delivery.
Fig.B: Functional and interoperability testing of Grandstream Networks DP752, DP730 & DP722
We plugged the PoE-enabled DP752 base into the Microsemi Digital
ceiling PoE switch, which provided both PoE and network connectivity. We
used the web interface for all the management and provisioning of the
DP752.
As shown in Fig.C, it presented us with tabular options such as: STATUS, PROFILES, DECT, SETTINGS, MAINTENANCE etc.
Fig. C: Grandstream Networks DP752 Base Station - Web Interface
In this test setup, the Patton SmartNode SN5570 eSBC will provide the
DP752 base station with SIP registrar services, SIP demarcation and
access to the SIP trunk in the cloud. We configured values relevant for
our testing by clicking on: Settings >> Network Settings >>
Basic Settings [IP Address mode, Preferred DNS Server, IP Address,
Gateway etc.] (refer to Fig.D). Then we moved to
Profiles >> Profile 1 >> General Settings [Profile Active,
Profile Name, SIP Server, Failover SIP Server, Prefer Primary SIP
Server, Outbound Proxy, Voice Mail Access Number] >> Network
Settings >> SIP Settings >> Audio Settings >> Call
Settings (refer to Fig.E). We clicked on DECT >>
General Settings [Base Station Name] Handset Settings [Handset Names]
>> SIP Account Settings [Account 1, SIP User ID, Authenticate ID,
Password, Name, Profile, HS Mode, Account Active] >> Handset Line
Settings [HS 1 Line 1 & HS 2 Line 1] etc. (refer to Fig.F).
We saved and applied all the changes that were made, then turned on
the DP730 handset and selected the Subscribe softkey and selected Base.
it started searching for the base station. In the interim, on the web
interface of the DP752 under Status, we clicked on the wireless icon for
Open subscription. It displayed the message, “subscription window
opened.”
The DP730 now showed us our base station named Juno and we selected
subscribe. It said easy pairing, and then the handset was subscribed.
We repeated this procedure for the DP722. The DP730 and the DP722 were
now successfully registered with the DP752 and ready to place and
receive calls as shown in Fig.C. The whole setup was
very streamlined just a few clicks on the relevant screens, thanks to
the well laid out DP752 web interface for configuration and management.
Fig. D: DP752 Base Station-Settings Page
Fig. E: DP752 Base Station-Profiles Page
Fig. F: DP752 Base Station-DECT Page
We tested the push-to-talk (PTT) feature of the DP730 and the DP722.
The DP730 already has a physical dedicated PTT key. For the DP722, we
programmed the left soft key as the PPT button. At the touch of a PTT
button, it enabled us to establish a PTT session between the DP730 and
the DP722 as shown in Fig.G
Fig. G: Display during a PTT session on the DP730 and DP722
We liked the PTT feature on the DP730 and the DP722, as we could
establish a speedy voice session with the other party without any need
for dialing, ringing and waiting for the called party to pick up and
answer. If a voice call was in progress, attempting PTT at that time did
not impact the voice call which was in progress. Further an incoming
voice call took precedence over our PTT session that was in progress.
We also turned the active SIP profile OFF so that the base now had no
SIP registration…however in this instance the PTT session continued to
work flawlessly.
To test the hunt group modes, we went to DECT >> SIP Account
Settings and for our account changed the HS mode to parallel mode,
circular mode and linear mode, each time the mode that was setup, worked
as expected. We were able to upgrade the software on the base station
and the phones wirelessly over the air as shown in Fig.H.
Fig. H: DP752 Base Station- Status Screen
As soon as you log into the DP752 web page, it shows you the status,
you are presented with information such as Account Status, DECT Base
Status, DECT Repeater status, Line Options, Network Status, System Info
etc. We could click on each of those respective tabs for further action
and more granular and useful information. From one easy to access
location these options gave us visibility and control into the
configurational and operational aspects of the DP752 and the subscribed
handsets. You could also click on actionable icons such as reboot,
unsubscribe individual/all handsets, page individual/all handsets etc.
Our functional and interoperability testing was successful,
demonstrating to us, that the DP752 VoIP base station along with the
DP730 and DP722 wireless phones are a feasible DECT solution to provide
mobility, good voice quality and security for our enterprise telephony
applications.
Conclusion
Grandstream networks DP730 and DP722 DECT wireless IP phones connecting
to the strategically deployed DP752 DECT VoIP base station will deliver
dependable wireless voice communications.
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