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What is ENUM? ENUM Syntax

What is ENUM? ENUM Syntax

ENUM (Telephone Number Mapping, E.164 Number to URI Mapping) is an addressing protocol that converts telephone numbers to URI format (name@domain). This allows you to access a SIP, H.323 or other Internet phone user by dialing a phone number.

The ENUM function aims to ensure that users can be accessed anywhere in the world with the same number, best quality and the cheapest way. ENUM maps a phone number to an Internet address in the DNS system. Thus, a user with an ENUM number can broadcast the DNS record to which the call will be routed. Even different routes can be defined for different types of calls (fax, video, etc.).

It is possible to obtain an ENUM record as if it were a domain name. Nowadays, you can obtain this registration free of charge through many registration services and VoIP service providers.

ENUM Syntax

ENUM allows normal phone (E.164) numbers to be displayed as DNS names ending in e164.arpa. A number can be decoded for one or more predefined services.

For example, a telephone number + 90-312-555-1234 will be displayed as 4.3.2.1.5.5.5.2.1.3.0.9.e164.arpa after issuing rules defined in RFC 3761 and below:

  1. All characters except digits are removed. (“+90-312-555-1234” becomes “903125551234″)
  2. A period (“.”) Is placed between each number. (“9.0.3.1.2.5.5.5.1.2.3.4”)
  3. The order of the numbers is reversed. (“4.3.2.1.5.5.5.2.1.3.0.9”)
  4. .e164.arpa is added to the end of the array. (“4.3.2.1.5.5.5.2.1.3.0.9.arp a”)

To respond to this syntax, the DNS server must have a record that looks like this:

$ ORIGIN 4.3.2.1.5.5.5.2.1.3.0.9.barley.
   NAPTR 10 100 "u" "E2U + sip" "! ^. * $! Sip: fatih.erikci@fatiherikci.com!" .
   NAPTR 10 101 & quot; u & quot; & quot; E2U + h323 & quot; .
   NAPTR 10 102 "u" "E2U + msg" "! ^. * $! Mailto: fatih.erikci@fatiherikci.com!" .

In this record you see three different routing sequences for the address 4.3.2.1.5.5.5.2.1.3.0.9. The first is SIP, the second is H.323 and the third is the SMTP response. Device selects which service to communicate by using these records.

How It Works?

The operating principle of ENUM is similar to the DNS queries we use on the Internet. DNS NAPTR resource records are used in queries.

ENUM Query and Call
ENUM Query and Call
  1. The phone calls an E.164 number (+90-312-555-1234)
  2. Gateway translates it (4.3.2.1.5.5.5.2.1.3.0.9.e164.arpa) and asks the DNS server.
  3. The DNS server responds to this query with a URI (sip: fatih.erikci@fatiherikci.com).
  4. The gateway sends the call to the SIP server as a SIP URI call.
  5. The SIP server rings the IP phone registered with the URI.
Cisco 730 Series Headset

Cisco 730 Series Headset

Cisco has also introduced 700 series headsets as well as new Webex series collaboration products at 2019 Partner Summit.

As you may know, Cisco has introduced many products and technologies in the field of unified communications and collaboration, and has entered the phone headset market with its 500 series headsets last year. This year at Partner Summit Cisco introduced the 700 series, which targets mobile workers segment.

Cisco Headset 730 Specifications

The 700 series is a series of headsets aimed primarily traveling mobile workers and 730 is the first model of the series. Especially in crowded environments, the noise canceling mechanisms ensure clear communication for both the headset and the called party. Below you will find detailed features of the Cisco 730 headsets:

  • Connectivity via Bluetooth 5.0, USB-A and 3.5mm
  • Premium Codec Support (SBC, AAC, aptX, aptXHD)
  • Active Background Noise Cancellation with 4 Microphones
  • Clear Voice Transmission with 2 Electret Condenser Microphone
  • Intelligent Sensor Technology (Mute on Earphone)
  • Boom-less Microphone
  • On-Ear Controls
  • Voice Activated AI
  • Cisco Headset App for Customized Experience (App Store & Google Play)
  • Automatic Firmware Upgrades

Here is a table that compares the Cisco 700 series headphones to the 500 series:

700 Series500 Series
Suitable For Mobile / Office Workers Office / Call Center Workers
Primary Connection Type Bluetooth 5.0 Wired/DECT Wireless
Talking Time15+ hrs9 hrs
Concurrent Connections 2 BT + 1 USBDepends on Base Station
Wireless Coverage65+ meters90+ meters
Ear Wearing StyleBinaural OnlyMonaural & Binaural
Color OptionsCarbon Black & PlatinumBlack
Active Noise CancellingYesNo
Noise Reduction MicrophoneYesYes
RJ-9 ConnectionNoYes
Cisco 730 Headset
Cisco 730 Series Headsets Have 2 Color Options

For more information about Cisco 730 series headsets, please refer to the datasheet page.

Price

The price of the newly introduced Cisco 730 series headset has not been set yet, but considering that the list price of the 560 series wireless headphones is around $600, my guess is that the list price will be in the range of $800 – $900.

Last Words

With the 730 series headset, Cisco has provided a professional solution for both the consumer market and business purposes. Cisco 730 headset is a very useful product especially for frequent travelers.

Cisco IP DECT Solutions

Cisco IP DECT Solutions

Cisco has introduced IP DECT Solutions, which have been developed to meet the need for wireless telephony. Until now Cisco didn’t have a DECT phone solution in its product family and this situation being complemented by Wi-Fi or 3rd party DECT products.

Cisco IP DECT products operate in the USA and Canada in the frequency band range of 1920 – 1930 MHz and 1880 – 1900 MHz frequency band in other countries. The IP DECT series currently consists of 210 series base station and the 6825 series handheld terminal.

Let’s take a closer look at these products:

Cisco 6825 IP DECT Handset

Cisco 6825 IP DECT Handset
Cisco 6825 IP DECT Handset
  • 2 Inches 240 × 320 Pixels 65K Color Display
  • Up to 17 Hours of Talk, 200 Hours of Standby Time
  • 2 Lines Supported
  • Narrow Band (G.726) and Wide Band (G.722) Codec Support
  • Bluetooth and 3.5mm Wired Headset Support
  • Waist Clip
  • Dimensions: 117 mm x 46 mm x 20 mm
  • Weight: 86 gr.

Cisco 210 IP DECT Base Station

Cisco 210 IP DECT Base Station
Cisco 210 IP DECT Base Station
  • 30 Handheld Terminal Registration Per Single Base Station
  • 5 Simultaneous Calls With Broadband Codecs and 10 Simultaneous Calls With Narrowband Codecs in Single Base Station
  • Multicell Technology
  • Scalability Up To 254 Base Stations
  • Total 1,000 SIP Device Registrations
  • Total 1,000 Simultaneous Calls with Broadband Codec, 2,000 Simultaneous Calls with Narrow Band Codec
  • G.711 (A-law & mu-law), G.722.2, G.726, G.729 (a & ab) Codec Support
  • PoE Class 2
You Can Use Multiple Cisco 210 Base Stations In Your Network (source cisco.com)
You Can Use Multiple Cisco 210 Base Stations In Your Network (source cisco.com)

In the Multicell concept, there is no central mechanism to control base stations. When deployed, one base station becomes the master station, and the other stations connect to master with the “Chain ID” defined on the base station and start their data synchronization. Yeah, that’s all.

Supported Platforms

Since the IP DECT series is a multiplatform product, it supports almost all PBXs using the standard SIP protocol. Here are Cisco’s officially supported platforms:

  • Asterisk
  • Cisco BroadSoft BroadWorks
  • Cisco BroadCloud
  • Centile
  • Metaswitch

Even looking at this list, you can actually understand Cisco’s strategy in unified communications solutions. 🙂

Last Words

Although Cisco has always put WiFi phones in the foreground, DECT phones -which have a soft belly in the projects- have started to meet the need. As I wrote in my previous article, we will see if Cisco can surpass its competitors in this field with OpEx oriented solutions such as UCaaS and HaaS.

What Is WebRTC?

What Is WebRTC?

We’ve been hearing the name WebRTC a lot lately. In fact, WebRTC, which has been in use since 2011, is not a new technology but is a technology that provides simultaneous media communication (audio and video). The most important feature of WebRTC, which has many advantages, is that it can work directly on many popular browsers without requiring additional software.

WebRTC stands for Web Based Real Time Communication. Multimedia applications can be designed using HTML5 and Javascript APIs.

We can define the communication format used in WebRTC as peer-to-peer. This communication is directly between peers, so you don’t need any media servers. WebRTC is free and has a BSD license, so you can develop WebRTC applications for free. (For example, you can experience a video conference virtual room with WebRTC at this link)

WebRTC Supported Browsers

Nowadays, the following browsers support WebRTC:

  • PC & MAC
    • Microsoft Edge 12+
    • Google Chrome 28+
    • Mozilla Firefox 22+
    • Safari 11+
    • Opera 18+
    • Vivaldi 1.9+
  • Android
    • Google Chrome 28+
    • Mozilla Firefox 24+
    • Opera Mobile 12+
  • iOS
    • MobileSafari / WebKit (iOS 11+)
  • Chrome OS
  • Firefox OS
  • BlackBerry 10
  • Tizen 3.0

WebRTC Components

There are 3 main components in WebRTC:

1. MediaStream API

The MediaStream API provides user access to the camera, microphone or screen using javascript.

2. RTCPeerConnection API

The RTCPeerConnection API provides NAT traversal, codec processing, mutual SDP negotiation, media transmission, and secure connection functions between peers.

3. RTCDataChannel API

The RTCDataChannel API provides the functionality of establishing bidirectional data transfer channels between peers.

Establishing Peer-to-Peer Connection

Signaling is a process that forms the connection between peers. It can be achieved by WebSocket, XMPP, SIP or any other mechanism. WebRTC technology utilizes protocols such as RTP, STUN, SIP and ICE.

WebRTC Signaling Process

Session Description Protocol (SDP)

Also known as SDP, it is a protocol used to communicate media capabilities (voice codecs, IP and port information, etc.) between peers before establishing a connection and to meet each peer at a common point.

Interactive Connectivity Establishment (ICE)

ICE is a framework for the NAT traversal mechanism. ICE collects all available candidates (local IP addresses, STUN return IP addresses, and transmitted IP addresses – TURN). All collected addresses are then sent to remote peers via SDP.

STUN Server

The STUN server enables peers to find public IP addresses, the types of NAT they use, and the relationship between the Internet-side port information associated with the local port information specified by NAT.

TURN Server

When STUN usage is not possible, it is used to transmit media streams over a TURN server (you may think of it as a proxy).

WebRTC is not always peer-to-peer (P2P), but in multiple communication situations (eg video conferencing), different solutions are available. Let’s take a look at these.

Multi-Point Communication Types

1. Mesh

In the mesh network, all peers send their streams separately to other connected peers directly on the network.

All Peers Communicate With Each Other in Mesh Topology

Since this structure is completely distributed, there is no need to have any media servers in the center. The disadvantage of the mesh structure is the use of high bandwidth. In a multi-video call using a mesh structure, if each user generates a 1 Mbps stream, the amount of data sent and received per user will be 4 Mbps in each direction.

2. SFU

SFU stands for Selective Forwarding Unit. An SFU receives incoming media streams from all users and then decides which users to send to.

SFU Transfers Media To All Peers Separately

In this model, each user transmits their own generated media stream to the SFU server. The SFU server can send whoever wants the stream. In this way, bandwidth is used more effectively. Similar with the mesh example above, if each user generates a 1 Mbps stream, the total outgoing data amount per user will be 1 Mbps and the total incoming data amount will be a maximum of 4 Mbps.

3. MCU

MCU stands for Multipoint Conferencing Unit. An MCU receives incoming media streams from all users, decodes them, creates a new layout, and sends it to all users as a single stream.

MCU Combines Media of All Peers & Sends a Single Stream to Peers

The difference of this structure from SFU is that a single combined stream will be sent to each user and the total transmission and reception amount per user will be 1 Mbps in each direction. The disadvantage of this structure, as you can imagine, is the high cost of the MCU with a high processing power in the center.

Initial Information About CUCM 14 & IP Phones That Won’t Be Supported

Initial Information About CUCM 14 & IP Phones That Won’t Be Supported

After version 12, CUCM will jump 13 for suchreasons, and will probably appear as 14 early next year. In version 14, as in the 11.5 and 12 versions, some IP phones will not be supported either.

IP Phone Models Won’t Be Supported in CUCM 14

With the CUCM version 14, some IP phone models that have been in the market for many years will no longer be available. The following list includes models that will not be supported:

  • Cisco Unified IP Phone 3911, 3951
  • Cisco Unified IP Phone 6911, 6921, 6941, 6945, 6961
  • Cisco Unified IP Phone 7906G, 7911G, 7925, 7925G-EX, 7926, 7931, 7936, 7937G, 7940, 7941, 7960, 7961, 7985
  • Cisco Unified IP Phone 8941

If you are using any of these phone models, CUCM will block registration for these phones after upgrading to CUCM version 14. If the IP phone stays on, it will create unnecessary network traffic as well as load the CallManager service as the phone will try to register itself again and again.

Reasons For Not Supporting These Phones

Apparently Cisco will only use the 7800 and 8800 series telephones at the end of the day, and the reasons for this are explained in four subheadings:

  • Security – Because older phone models cannot receive critical software updates, to protect users as new security issues arise.
  • New Features and Applications – Since some phone models have been introduced many years ago (eg, 7900 series was introduced in early 2000’s), the older processors and hardware in these phone models creates a number of problems in implementing new applications and security features, so that the user experience can not be maximized.
  • Sustainability – To ensure sustainability because older phones no longer have development support or regression tests.
  • End-of-Life Products – The phone models listed above have made end-of-sale and end-of-life announcements in the past. These phone models will expire when the CUCM 14 is launched.

If you have a plan to upgrade to CUCM 14 and you have older model phones that is no longer supported, Cisco offers the following advice:

  1. Identify unsupported phones and schedule your upgrade to version 14 on time.
  2. Take advantage of competitive trading programs (Trade-in, Cisco Capital, etc.) or work with Cisco teams to cost-effectively renew the phones.
  3. Consider the strategy of migrating to software-based end devices like using Jabber.
  4. Talk with Cisco partners about switching to 7800 and 8800 series IP phones which are supported.

You can find the Field Notice on Cisco.com by clicking this link.