STIR/SHAKEN 101

STIR/SHAKEN is a technology framework designed to reduce fraudulent robocalls and illegal phone number spoofing. STIR stands for Secure Telephony Identity Revisited. SHAKEN stands for Secure Handling of Asserted information using toKENs. Its goal is to prevent fraudsters from scamming consumers and businesses through robocalls and illegal phone number spoofing, while making sure that legitimate calls reach the recipients.

The FCC has adopted rules requiring service providers to deploy a STIR/SHAKEN solution by June 30, 2021.

STIR/SHAKEN is a great tool to help providers restore confidence in the calls they're connecting. It is not a one-stop solution for preventing telecom fraud. In fact, its main use is to verify that a call being made is in fact from the owner of the telephone number.

To verify the ownership of the phone number, there are 3 types of attestation:

1. A (Full) - The service provider knows the customer and their ownership to use the phone number.
2. B (Partial) - The service provider knows the customer, but not the source of the phone number.
3. C (Gateway)  - The service provider has originated the call onto the network but can’t authenticate the call source and phone number. e.g international gateway.

On the terminating end, attestation of A and B are usually considered good calls but not guaranteed from being blocked, because analytic engine and fraud system have other factors considered than attestation result.

SIP main functions

1. User location - locate the end user geographically
2. User availability - available, busy, DND presence infomation
3. User capability - determine the media being used and parameters associated with the media
4. Session setup - establish session parameters for both caller and callee
5. Session management - transfer a call, end a call, or modify session paramaters

https://tools.ietf.org/html/rfc3261

Telephony terminologies

POTS (Plain-Old-Telephone-Service) was created in 1876.

ISDN (Integrated Services Digital Network) was introduced in 1988. ISDN comes in two forms: the basic rate interface (BRI) and the primary rate interface (PRI)

PRI can transfer more data, making it easier to transfer things like HD audio and video and more suitable for enterprises.

PSTN (public switched telephone network) is simply the global aggregate of all these interconnected copper telephone systems.

SIP (Session Initiation Protocol) was introduced in 2000.

Session Initiation Protocol is a set of communication standards that allow (for the most part) the setup and termination of voice or video calls. SIP allows voice traffic to be carried over data networks, including the internet. SIP is considered a type of VoIP.

VoIP (Voice over Internet Protocol) is an overarching term for the technology included in all IP based telephony.

Over-the-Top (OTT) VoIP, services such as WhatsApp, require both calling parties to have an active data connection and carry calls entirely over data networks.

The beauty of SIP is that it can be used to send calls to and from the PSTN, using media gateway.

A SIP Trunk is used to transfer a call between its origin and destination using the Public Switched Telephone Network (PSTN) or in the case of a Voice over Internet Protocol (VoIP) call, the internet. It describes the process of allowing multiple callers access to the same telephone service by sharing a line that can handle multiple calls instead of providing an individual line for each call.

Understanding Audio Quality

When referring to audio quality, bitrate is a measurement of bits per second that audio distributes. The sound quality will improve as the bitrate improves. For example, MP3 files with a bitrate of 128 kbps are more likely to sound better than MP3 files with a bitrate of 64 kbps.

Digital audio has a sample rate, bit depth and bit rate. They are usually compressed to reduce file size and stream more efficiently over networks. Compression can be lossy or lossless.

1. sample rate - the number of audio samples captured every second. Telephone networks and VOIP services can use a sample rate as low as 8 kHz.  
2. bit depth - the number of bits available for each sample. The bit depth may be 8-bit, 16-bit, 24-bit, 32-bit. The higher the bit depth, the higher the quality of the audio. Bit depth is usually 16 bits on a CD and 24 bits on a DVD. 
3. bit rate - the number of bits encoded per second of audio, or the number of bits transmitted or received per second. Bit rates are usually measured in kilobits per second (kbps).

Bit rate calculation
bit rate = bit depth * sample rate * number of channels

File size calculation
http://www.audiomountain.com/tech/audio-file-size.html

Uncompressed - Lossless (Audio CD, PCM WAV, AIFF)
Lossless are the highest quality files you can get and come mainly in the form of WAV (Microsoft), AIFF (Apple) & FLAC.  These files start off at the equivalent quality of a CD with a bitrate of 1411 kbps and a sample rate of 16bit but can go all the way up to 24bit / 192Khz. A WAV for example can be approximately 3.5 times bigger than a 320kbps MP3.

FLAC and ALAC are open source lossless compression formats.

Compressed - Lossy (mp3, m4a, aac, wma, ogg)
Compressed files come in varying quality rates and formats of which MP3 & M4A are the most popular. The bit rate for compressed files can go from 8 kbps up to 320 kbps.

Ogg Vorbis is an open source alternative for lossy compression.

Kari's Law Compliance in 3 sentences

1. No more outside line prefix when dialing 911.
2. Notifications to designated personnel when a 911 call has been made. These notifications can take the form of a phone call, email, SMS/text message and etc.
3. Location info to PSAP (public-safety answering point) needs specific address info like building, floor, suite, and even conference room info. (Complementary to Kari’s Law, RAY BAUMS Act)

Why SBC in Cloud PBX?

A session border controller (SBC) is a dedicated hardware device or software application that governs the manner in which phone calls are initiated, conducted and terminated on a Voice over Internet Protocol (VoIP) network. Phone calls are referred to as sessions.

An SBC acts a router between the enterprise and carrier service, allowing only authorized sessions to pass through the connection point (border). The SBC defines and monitors the quality of service (QoS) status for all sessions, ensuring that the callers can actually communicate with each other and that emergency calls are delivered correctly and prioritized above all other calls. An SBC can also serve as a firewall for session traffic, applying its own quality of service (QoS) rules and identifying specific incoming threats to the communications environment.

SBCs are inserted into the signaling and/or media paths between calling and called parties in a VoIP call, predominantly those using the Session Initiation Protocol (SIP), H.323, and Media Gateway Control Protocol (MGCP) call-signaling protocols.

The term “session” refers to a communication between two parties – in the context of telephony, this would be a call.

The term “border” refers to a point of demarcation between one part of a network and another.

The term “controller” refers to the influence that SBCs have on the data streams that comprise sessions, as they traverse borders between one part of a network and another.

SBC major features:

• SBC can act as a SIP firewall
• SBC can do call admission control
• SBC can perform network addresses translation (NAT)
• SBC can adapt to different SIP messages forms
• SBC can do call recording using software
• SBC can do transcoding (H.323/SIP, G.729/G.711, IPv6/IPv4, SRTP/RTP, TLS offloading)
• SBC can route SIP sessions (ingress/egress point for SIP trunks)

Sonus SBC can do the followings in terms of security:

• IP Trunk Groups
• Call Admission Control
• Rate Limiting
• Access Control Lists
• Traffic Policing
• IPsec
• Topology Hiding
• Split DMZ
• TLS Signaling
• SRTP
• Firewall/NAT
• DoS/DDoS Protection
• SIPS
• Dynamic Blacklisting
• Rogue RTP Protection
• Encrypted Communication
• Malformed Packet Protection

Contact Centers

Cloud-based Contact Center (CC) is getting very popular with a few major players in 2019. Almost each Contact Center can provide essential call center features, as well as Omni-channel routing, integrations, analytics, live reporting, workforce optimization.

Five9
Five9 has Predictive AI technology, with features such as intelligent call routing, dialer modes, CRM integration (Salesforce, Zendesk), analytics, workflow management, and an omni-channel solution.

Talkdesk
Talkdesk provides call center features such as ACD, IVR, dialers, CRM integrations, real-time reporting & analytics, workforce management, and AI automation.

Genesys
Genesys call center software is powered with modern features ACD, IVR, routing, workforce optimization, and omnichannel support.

InContact
NICE inContact’s CXone platform comes packed with features such as omni-channel routing, analytics, workforce optimization, integrations, automation, and AI, all built on an open cloud foundation.

Twilio
Twilio platform is highly customizable with communication APIs for SMS, voice, video & authentication. Twilio Flex is the first fully-programmable contact center platform.

8x8
8x8 supports features such as omni-channel routing, IVR, integrations, analytics, supervisor management systems, and agent productivity knowledge. The ultimate plan comes with a full list of features, including a multichannel contact center, advanced analytics, and predictive dialer. 

RingCentral
RingCentral CC has features such as omni-channel routing, CRM integrations, reporting & analytics, and agent management software that allows businesses to build a powerful customer engagement platform. Its ultimate plan supports advanced IVR and ACD, as well as omni-channel capabilities that supporting things like chat, email, SMS, and social media.

Please note that RingCentral and 8x8 are also major cloud PBX players in the industry, besides Cisco and Microsoft calling.

https://getvoip.com/blog/2019/07/12/top-call-center-software-pricing/

Australian IPND

The Australian Communications and Media Authority (ACMA) is an Australian government athority.

Integrated Public Number Database (IPND) is an industry-wide database containing all listed and unlisted public telephone numbers. It is managed by Telstra.

Carriage service providers (CSPs) that supply a carriage service to an end-user of a public number must provide the public number and the associated customer data to the IPND Manager.

Where a customer’s IPND record is inaccurate, the CSP must correct the data. This is subject to CSPs checking that the requested changes comply with their regulatory obligations - for example, the CSP should cross-check a customer request to change the spelling of a name against a form of identification.

The ACMA is responsible for monitoring and enforcing CSP compliance with the obligations to provide accurate and timely customer information to the IPND. A key aim of the IPND compliance program is to improve the quality of data in the IPND. This is important because the harms that can result from inaccurate data in the IPND can be serious.

https://www.acma.gov.au/Industry/Telco/Numbering/IPND/integrated-public-number-database-numbering-i-acma

Voice quality factors

There are the following main factors impacting voice quality. Focusing on these improvements, it will help a lot on voice quality.

BANDWIDTH
It is the rate of data transfer, bit rate or throughput, measured in bits per second (bit/s).
It depends on the codec—how the data is compressed to be sent and received. When analog voice is digitized, if it is sampled 8,000 times per second. Each sample is encoded in 8-bits. So, we need to have a bandwidth of 64,000 bits per second (or 64 kbps) one way to send that voice data.

PACKET LOSS
Packet loss occurs when one or more packets of data traveling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion. Packet loss is measured as a percentage of packets lost with respect to packets sent.

You know when the voice drops in and out randomly on a phone call? That’s packet loss. Packets can be dropped randomly while files are being delivered from one phone to the other, resulting in those blank spots you might have experienced. The good news here: today’s IP network equipment is super reliable and even at a loss of around 3%, the quality that listeners hear is still better than what you’d hear on an average cellphone.

JITTER
A network with constant delay has no packet jitter. Packet jitter is expressed as an average of the deviation from the network mean delay.

Everyone’s been there—you’re on the phone and there’s a bit of a delay, so you keep talking over each other and finally have to take a pause or hang up. That’s jitter at work—another network enemy. Jitter occurs when packets are delivered across the network at delayed or inconsistent times. Some providers tackle this problem by introducing jitter buffers—essentially an agreed-upon delay. Jitter buffers can be a double-edged sword as a big buffer will introduce latency, and a very small one might not be effective.

LATENCY
Network delay is an important design and performance characteristic of a computer network or telecommunications network. The delay of a network specifies how long it takes for a bit of data to travel across the network from one node or endpoint to another. It is typically measured in multiples or fractions of seconds. Delay may differ slightly, depending on the location of the specific pair of communicating nodes.

Latency is introduced when voice data travels from point to point across the network—the more endpoints or other networks your call has to cross, the more latency you’re likely to experience. A network’s on-net infrastructure can help you minimize latency. Staying on-net means the call doesn’t have to bounce from carrier to carrier to be completed. The less hops a call has to take, the better the call will sound.

Line Information Database (LIDB)

A Line Information Database (LIDB) is a record kept by the local telephone service provider that consists of a number of details for each customer. The categories included in a Line Information Database include subscriber information, such as a service profile, name, address and credit card validation information. For instance, caller ID name is stored in LIDB. The acronym LIDB often is pronounced "lid-bee."

LIDB is used in the United States and Canada by traditional telephone companies to store and retrieve Caller ID records. Local phone switches, also known as Class 5 switches, use SS7 signaling protocol to query these centralized databases and pass this information during call set up. The information includes subscriber's information such as service profile, 10-digit line number, service provider ID, equipment indicator and billing specifications.

Internetwork Calling Name (ICNAM) is a service that works in the same manner as LIDB. With ICNAM, rather than returning calling card or bill name validation data, the query data returned is the name of the calling party.

There are two main components to the Caller ID info seen by a receiver of a call: the number and the name. The number is the "easy" part, but the name takes a little more effort to set up.

• Setting your own CNAM within your PBX has no effect in the U.S. and many other countries. Only the CID number is transmitted to the terminating carrier, and those carriers then look up the name from a database.

• If you will be making calls to Canadian numbers, you can pass the Caller ID Name from your PBX.

In the U.S., CNAM information depends on what is stored in a Line Information Database (LIDB), which contains records of numbers and names. There are many different databases, and each carrier decides which database to use.

UCaaS security framework

Cloud UCaaS (Unified Communications as a Service) needs a security framework to make it secure and reliable in the cloud for Meetings, Phone and Chat.

1) Secure data center
UCaaS provider (vendor) needs facilities with strong physical protections, redundant power, and tested disaster recovery procedures.

2) Robust network security
UCaaS vendor must add unique protections designed to prevent attacks on the infrastructure, preventing service disruption, data breaches, fraud, and service high-jacking. Also needs to resolve firewall traversal problems in VoIP systems with network address translation (NAT) support for static IP configuration and “Keep-Alive” SIP signaling.

3) Secure voice
All voice traffic within cloud phone system should be encrypted to prevent eavesdropping on voice calls.  Provide additional security for IP phone calls using SIP over TLS and SRTP encryption.

4) Data encryption
All data should be encrypted in transit and at rest, with audit-able record-keeping and reporting. It includes everything from physical protections at data centers to encrypted storage to comprehensive digital tracking with clear audit trails.

5) Fraud prevention
The service provider should have protections built in to the service layer and should conduct continuous monitoring for dangerous anomalies or other indicators of toll fraud and service abuse.

6) User access controls
To ensure only authorized users access cloud communications accounts and services, the vendor should implement at a minimum strong password policies and ideally two-factor authentication as well as single sign-on (SSO).

7) Account management and administration
Administrators can instantly revoke the remote user’s access to the cloud network—and thereby to customer contacts, CRM info, and other corporate information—and almost no data resides on the device itself.

Hunt Group vs Call Queue

Hunt group and call queue have the same purpose to distribute incoming calls. They are very similar. The key difference is queue. However, call queue has more complexity,  better customization, better management and better distribution, and is more expensive.

In telephony, Hunt group (a.k.a. ring group, call group, line hunting) is the method of distributing phone calls from a single telephone number to a group of phone lines. Specifically, it refers to the process or algorithm used to select which line will receive the call. Hunt groups are supported by most PBX phone systems.

A hunt group is an extension or phone number that rings directly to multiple phones defined in the Hunt Group. Calls can be delivered in several different manners. They can be sent sequentially, (i.e. if nobody answers it goes to the next line in the group), or simultaneously, (i.e. all the phones ring and the first to answer takes the call). Other options include, round robin, percentage weighting, least used line etc.

When a call comes into a ring group, it rings the phones in the group as per the setup. But if a phone is busy then that phone is temporary out of the group. When all phones in the group are busy then the ring group will no longer "hold" the calls and will follow the call handling rules set up in the hunt group for when the phones are busy/not answered. Usually the call is forwarded to voicemail or gets disconnected.

When a call comes into a call queue, it will keep the caller in the queue even if ALL phones in the queue are busy. The queue will only move on when the queue timeout is reached or there are no extensions logged in.

Call Queue is a concept used in inbound call centers. Call centers use an Automatic Call Distributor (ACD) to distribute incoming calls to specific resources (agents) in the center. ACD holds queued calls in First In, First Out order until agents become available. Administrator can monitor call queue status and make dynamic adjustment to the queue based on incoming call volume.

Usually there are 4 Types of Call Queues

1. Round-robin (longest idle) – routes callers to the available agent that has been idle longest.
2. Ring All –  routes callers to all available agents at the same time.
3. Linear Hunt – routes callers to the available agents in a predefined order. The order is defined when editing the queue's agents.
4. Linear Cascade – routes callers to groups of available agents in a predefined order. The order is defined when editing the queue's agents.

To sum up, if you want to keep your callers in the queue while you finish your other calls then use a queue. If you want them to go to voicemail then use a hunt group.