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What Is SIP ALG and Why VoIP Users Should Disable It

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In this post, we discuss what SIP ALG is and how it can affect the quality of your VoIP phone calls. Disable SIP ALG to improve VoIP call quality and ensure there are fewer interruptions.

What is SIP ALG?

Session Initiation Protocol (SIP) is an internet protocol with voice data packets that initiates, maintains, and terminates voice communication between two users. SIP is used for voice calling over LTE and VoIP phone systems.

Routers used to connect to the internet also segment the provider and your internal network through Network Address Translation (NAT). This is to add an additional layer of security through a firewall allowing only authorized systems access as they connect with a network’s computers and devices.

The main purpose of SIP ALG — Application Layer Gateway — is to prevent problems caused by a router’s firewall. ALG prevents these issues by keeping an eye on the VoIP traffic (voice data packets mentioned earlier) and modifying them, when necessary. ALG works as a proxy to rewrite the destination for these packets. By doing this, ALG can improve connectivity.

Why VoIP Users Should Disable SIP ALG

Many routers have the SIP ALG feature turned on by default. With this feature on, VoIP traffic (voice data packets) can get lost due to router firewalls when transferred between the phone and the VoIP provider.

And because of this, it can lead to multiple VoIP problems, including:

  • One-way audio
  • Phones not ringing on incoming calls
  • Calls sent directly to voicemail, especially when not set to do so
  • Dropped calls, even after connecting

This is why one of the best ways to improve VoIP call quality, among others, is to disable the SIP ALG feature.

How to Disable SIP ALG in your VoIP System

Disabling SIP ALG is quick and easy, and depends on the type of modem your business uses. For most routers, you will need to:

  • Log into your router’s control panel.
  • Navigate to Advanced or Security settings.
  • Locate SIP, ALG, or Firewall settings (depends on your router’s set-up).
  • Uncheck the SIP or ALG box.
  • Save and reboot/restart your router.

If your router’s settings are not as clear, you can always reach out to your provider and ask for specific instructions.

Protect and Maintain VoIP Call Quality

Disabling SIP ALG is a common way of troubleshooting VoIP issues. However, there are other VoIP call quality issues such as jitter, packet loss, and latency that can affect the way your business communicates with its customers. Most of these issues stem from low-quality internet or insufficient bandwidth. Speak with our representatives today to learn how your internet bandwidth can affect your VoIP phone system. Call us at 1 (877) 898 8646 or chat with us online today.

What is an Auto-Attendant and 3 Benefits of Using One

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Need an automated telephony service to help you manage business calls? Learn how an auto-attendant works and how your business can use one effectively.

What is an Auto-Attendant?

Auto-attendant refers to a telephony service wherein a voice menu system answers incoming calls and transfers callers to the appropriate extension without the help of an operator or receptionist. Other terms for auto-attendant include automated attendant, digital receptionist, and interactive voice response (IVR).

How Does an Auto-Attendant Work?

The automated attendant can be understood as an automated call answering system that helps transfer incoming calls while providing callers with general information about the company and its services. But the main point of an auto-attendant is to manage incoming calls effectively so callers are not left confused or arrive at the wrong agent or department. An auto-attendant may have the following features to ensure better call management:

  • Greeting messages
  • Business information (office hours, location, list of services, etc.)
  • Automated company directory (with extensions for users and employees)
  • Call transfer and routing options
  • Menu prompts such as Repeat, Exit, Speak to Representative, Operator

Auto-Attendant vs IVR: What is the Difference?

The terms auto-attendant and IVR are often used interchangeably. However, there are a few differences between these services. The main difference is that the interactive voice response system is a more advanced system with additional features.

Auto-attendants route and transfer incoming calls so customers don’t wait in queue for long. And if waiting is required, then hold music is played. IVR systems include more smart features. For instance, IVR systems have voice recognition that enables callers to speak or say what they need instead of pressing a button on their keypad. This way, callers are not limited to the set menu available and can explain the reason for their call better.

Additionally, the IVR system can collect information about the customer and route calls accordingly or inform the agent beforehand. This includes account numbers, customer IDs, and so on. As such, the IVR can prepare the appropriate agent before they proceed to assist the customer. Furthermore, the IVR’s self-help menu allows for callers to complete certain actions and tasks without needing an agent. For example, the IVR can assist callers in paying bills, checking one’s account balance information, scheduling appointments, and so on.

3 Benefits of Using an Auto-Attendant

So, what does an auto-attendant do and how can your business benefit from such a service? Automated attendants or IVR systems have countless benefits that support call management and improve caller experience. Here are some of the top benefits of using an auto-attendant:

1. Effective Call Management

Since calls can be automatically distributed based on set rules, callers reach the appropriate department or agent quickly and accurately. This is especially useful for businesses that have large call volumes and struggle with answering calls effectively.

2. Increased Productivity and Efficiency

You can study your customers’ needs and preferences and customize your auto-attendant to provide them with more useful options and reduce wait times. Additionally, you will also reduce the number of times agents receive calls that are not related to their department, increasing efficiency and productivity.

3. Cost Savings

With smart call routing, you will not need a secretary or receptionist working to manage your calls. Your company can save on hiring extra staff and place more emphasis on improving customer experience.

Where Can I Get an Auto-Attendant?

You can get an auto-attendant from virtual phone number providers like United World Telecom. Speak with our experts to learn more; call us at 1 (877) 898 8646 to get started today!

Troubleshooting the 7 Most Common VoIP Issues

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Having VoIP problems and don’t know how to solve them? Here we go over troubleshooting for the 7 most challenging VoIP issues.

7 Common VoIP Troubleshooting Problems

VoIP phone systems help businesses save about 50%-75% of communication-related costs. This is because such systems offer flexibility, mobility, and scalability which helps users connect from any location and communicate through advanced technology.

However, even VoIP phone systems — with their advanced features, high voice quality, and more — are not devoid of possible quality issues. Thankfully, most VoIP call quality can be improved without IT help so you can continue communicating effectively.

Here are the most common VoIP issues and a simple guide to troubleshooting them.

1. Inability to Make Calls from a Device

Struggling to make VoIP calls or outbound calls from your device? An inability to make calls can be due to a failure to connect, inadequate internet support, and more. For some businesses — like a call center — not being able to make outbound calls to customers and leads can essentially shut the business down until you find a solution.

Most likely, the cause of this problem is the SIP ALG feature is turned on, on your router. Session Initiation Protocol Application Layer Gateway (SIP ALG) is a common feature in commercial routers and is turned on by default. The main task of a SIP ALG is to reduce or prevent issues resulting from router firewalls. It does so by constantly inspecting your VoIP call traffic. However, SIP ALG may modify packets (voice signals) in unexpected ways, leading to problems such as incoming and outbound calls failing and phones not registering.

Solution: A simple solution for outbound VoIP calls failing would be to turn off the SIP ALG feature. If you still experience the issue, then try repositioning the VoIP devices onto a VLAN.

2. Dropped Calls

One of the most common VoIP problems is dropped calls. This causes a lot of frustration, especially during business calls. This is when the call suddenly ends mid-conversation without the speakers hanging up. Call centers or large enterprises with large call volumes face this issue the most.

Solution: First, ensure all devices, software, and hardware associated with your VoIP phone system are updated and running on the current version. If you are still experiencing the issue, disconnect all devices and turn them back on one at a time. This may be time-consuming but it will help you identify exactly which device is the root cause of the problem. Speak with your small business VoIP provider if you notice that calls get dropped after a certain amount of time. They may have an automatic disconnect feature to ensure calls are not left open by mistake.

3. Jitter

Jitter is one of the most common VoIP problems. Jitter directly affects voice quality and communication, leading to jumbled, muffled, or missing audio. As voice data packets travel from one destination to the next, some packets may arrive before the other. This leads to out-of-order or missing parts. If such voice quality occurs for more than 30 milliseconds then the overall call quality is impacted. As such, when finding a new provider, look for one that can keep the delay under 20 milliseconds.

Solution: Your internet may not have enough bandwidth for VoIP. Upgrade your internet connectivity by contacting your ISP.

4. Echo

This is a pretty straightforward VoIP concern. Telephone echo leads to voices being repeated at various points, leading to confusion and possible miscommunication. Often the recipient of the call hears the echo while the caller may or may not be aware of this VoIP problem. Echo can be a result of either feedback during the conversation or a VoIP phone system issue. As such, it can be troublesome when conducting important business calls such as conference, sales, and support calls.

Solution: First, if your phone is using the speaker option, take the call off the speakerphone. When using a speakerphone, the voice has to travel through multiple microphones and speakers leading to disruption in the audio for the recipient. Additionally, you may even need to test the phone headset you use and consider getting a high-quality replacement. Lastly, echo can be a result of a bad internet connection or inadequate bandwidth. Check your speed with an Internet speed test and also reevaluate your wall jacks, Ethernet cords, and other cables to ensure there are no damages.

5. Broken/Muffled Audio

Broken, muffled, or choppy audio refers to words and audio being dropped, interrupting calls when connected. This is one of the most common VoIP issues users face. Thankfully, it has a solution.

Solution: How you solve the problem of broken audio depends on who is experiencing it. If your business is experiencing the issue, it is most likely due to insufficient bandwidth that leads to packet loss as all voice packets are transferred successfully. A common VoIP troubleshooting solution for this problem is to turn off other applications that take up a lot of network space and are not needed for business. This includes streaming services like YouTube or Netflix and so on. Additionally, make sure your router’s Quality of Service (QOS) settings have the VoIP service on priority.

6. No Sound

Similar to broken audio, a voice call with no sound after connecting can lead to frustration and interruptions in communication. No sound in a voice call can be a one-way issue (where one party hears but others can’t) or a two-way issue (where both parties cannot hear).

Solution: One reason for a lack of sound during calls may be because of firewalls blocking RTP packets. Examining and possibly disabling your SIP ALG can solve this problem.

7. Phone Doesn’t Ring on Incoming Call

This VoIP issue is pretty straightforward: missing calls from important customers and clients because the phone doesn’t ring. Another version of this issue is if your calls are sent directly to voicemail instead of an employee.

Solution: Thankfully, this common VoIP problem has an easy solution. First, ensure your device is registered within your VoIP phone system and VoIP provider. Also, check to make sure your device is not on the Do Not Disturb setting and has the correct call forwarding settings and configurations.

Choosing a Reliable VoIP Provider

Need a new VoIP phone system that comes with high call quality and easy problem-resolution? Finding a reliable VoIP provider for your small business can be tough if you do not know what to look for. United World Telecom has been in the business for over 25 years and we offer top-quality communication services. Try our VoIP phone system solutions! Call us today at 1 (877) 898 8646 or chat with us online!

What is a RespOrg?

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A quick guide to what RespOrgs are, how they work, and how businesses can benefit from using a RespOrg service provider for their toll free numbers.

RespOrg: Definition

A Responsible Organization or RespOrg is a company (usually a telephone company) that is certified to have access to a centralized database of toll free numbers. This centralized database is known as the 800 Service Management System or SMS/800.

How do RespOrgs Work?

RespOrgs are in charge of managing toll free databases, assigning numbers, and keeping records. If you want a toll free number for your business, you will need to contact a RespOrg. For customers, RespOrgs come into play when porting a toll free number. To port a toll free number, a current user will have to change the RespOrg ownership from the old carrier to the new carrier. You will need a Letter of Authorization from your new carrier and your current RespOrg must authorize the release of the number to the new RespOrg or carrier.

A business that has high toll free traffic can take advantage of one of the below choices:

  • Become their own RespOrg
  • Use a single carrier for all of their call volume
  • Use a RespOrg service provider

Port your toll free number to United World Telecom.

How to Become a RespOrg?

RespOrgs can be large or small companies or even run by a solo business owner. Some toll free number carriers or business phone service carriers may also be RespOrgs. Currently, the US has about 400 RespOrgs. To become a RespOrg, a business goes through a certification process.

Technically, any company or organization that uses a toll free number can become a RespOrg. To become a RespOrg, your business will need to do the following:

  1. Complete and submit a ten-page service establishment form
  2. Pay a deposit (avg. $4000)
  3. Pass a certification exam to be certified

Should My Company Become a RespOrg?

While becoming a RespOrg is an easy process, there are a few factors to consider. For example, you will need to factor in the salary of the employee managing the toll free traffic. The cost of being a RespOrg for your business — as opposed to using a RespOrg provider — may entail increased expenses. Plus, if your employee leaves, you will need to train and certify a new employee, which will require additional costs.

Many businesses, therefore, choose to work with a RespOrg service provider to reduce costs. RespOrgs will work with your business and your specific needs to offer you the best pricing. Some benefits of using a RespOrg include:

  • Ability to route calls to different carriers
  • Routing calls at different times of the day
  • Taking advantage of low-cost carriers in different countries
  • Access to Disaster Recovery — in case your toll free carrier is shut down, traffic can be routed to a secondary carrier

A company with high toll free traffic will find it beneficial to utilize a RespOrg service provider instead of becoming their own RespOrg.

Get a Toll Free Number for Your Business Today!

United World Telecom offers toll free business numbers for more than 160 countries around the world. You can get a toll free number to enter new markets and extend sales and customer support services to more customers. We also offer number porting services for businesses that currently have a toll free number but are not satisfied with their service. Sign up on our website today or call us to learn more!

5 Signs You Need a New Business Phone System

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Your business phone communication system can make or break your sales and customer support efforts. More specifically, a bad phone system can lead to miscommunication and ineffective communication which can result in a loss of valuable customers and leads. Here we look at 5 signs that scream “Your business needs a new business phone system!”

Finding the Right Business Phone System

Since so much of your business’ success depends on good communication, it is important to find the right business phone service. Here are 5 signs you need a new business phone system. In other words, consider getting a new phone system if your current office communication system has any of the following:

1. Bad Pricing and Low Reliability

Are you struggling with paying the bill for your current phone system? Is the pricing not straightforward, leading to confusion and paying more than you originally signed up for? Your business phone system provider should be reliable and trustworthy. And you should be able to predict your monthly bill without any surprises for better accounting.

Switch to a business phone service provider that helps you understand what to expect from your monthly bill. As you sign up, you should have a clear idea of what you are paying for. This means no hidden fees or set-up/installation fees that just turn up out of nowhere. With a predictable monthly bill, your business can manage its communication-related finances better.

Related: How To Set Up a Business Phone System in your Office

2. Less Variety of Virtual Communication Features

With the advancements in telecommunications and technology available, businesses now have access to many opportunities and top-notch services and features. And so, if your current phone system is offering you the bare minimum, then you are missing out.

Advanced services such as global call forwarding, call recording, a variety of virtual numbers, and so on, can help with effective call management by providing your business with the right communication tools. These services can improve productivity and efficiency in your office which will lead to better customer service and more sales.

3. No International Call Forwarding

Any business with a goal for expanding globally must have international call forwarding. An international call forwarding service enables businesses to connect easily with their global customers. With this service, you can have incoming calls from different countries route to a destination of your choice. For example, your US-based company can receive international calls from customers in the UK, Asia, Australia, and more via virtual phone numbers. These calls are forwarded to your main office in the US or remote offices around the world via international call forwarding.

As such, call forwarding helps businesses maintain global connectivity by providing customers around the world with inexpensive ways to call for product inquiries or customer support. Even if your business is located outside the country, customers can call your local or toll free number and evade long-distance calling rates. This encourages customer calls and builds trust and credibility for your international business.

H3: 4. Limited Customer Support

Customer support is an essential tool for any business. And so, if you cannot reach customer service for help with your business phone system, especially during emergencies, then again, you stand a chance of losing out on customers. Unreachable or bad customer service can break a company.

Look for a business phone service provider that is available 24/7 and offers multichannel support. For example, United World Telecom offers 24/7 customer support via voice, live chat, email, and trouble tickets. We also have an online knowledge base with self-service information.

5. Long Terms Contracts

Lastly, you want a business phone system that does not force you into long-term contracts. In case things change within your business, you should be able to scale up or down or switch to new services. A provider that locks you in with contracts or high cancellation fees can make it difficult to grow your business.

Why Should You Consider United World Telecom?

Since 1996, United World Telecom has been providing businesses around the world with business phone communication systems and virtual communication tools to boost sales, customer support, and international business. Advanced services and features we offer include:

  • International call forwarding
  • Outbound calling with customizable caller ID
  • Hosted call recording
  • Cloud IVR
  • Extensions and DID numbers, and more

United World Telecom’s plans are straightforward with no hidden fees or long-term contracts. We don’t force our customers into commitments as we know our service will speak for itself. We offer five different plans for businesses of every size and type; so you can choose the plan that works best for your communication needs.

Get a New Business Phone System Today!

We offer a variety of services that can help you create the ideal business phone system for your specific company. Get a VoIP business phone system today by calling us for more information or by signing up on our homepage!

7 Advantages of Using Automatic Call Distribution

Businesses the world over have been using an Automatic Call Distribution (ACD) system for better call management and to enhance caller experience. Let’s review the top 7 ACD advantages to understand how your business can use an ACD system to improve customer service.

7 Automatic Call Distribution Benefits

An ACD system is a telephony service that automatically routes incoming calls based on rules input previously by the account manager. These rules are based on various factors such as the time of the call, location of the caller, agent skills, agent history, and more. By routing calls automatically, the ACD system assists businesses by sending callers to the right agent or department for customer support or sales.

Read on to learn about the top 7 ACD advantages.

1. Automatic Call Routing

The most attractive benefit of an ACD system is its ability to route calls automatically and intelligently. The calls are routed based on predetermined rules and algorithms. Some ways to use ACD for call routing include:

    • The caller or customer’s information and history with the company
    • The caller’s area code or location
    • The time of the call
    • Agent availability
    • Agent skill such as language or area of expertise
    • Voice menu configurations

By using these call routing strategies, your business can save on missed or dropped calls, wrong transfers, and wasted time. Instead, callers will reach the right agent or department quickly, leading to better customer service and call resolution.

2. Quick Response to Calls

By transferring calls immediately to the right department and the right agent, your employees can answer customer calls quickly, almost instantly. Furthermore, you can even provide users the ability to leave a voicemail or offer a callback option during high call traffic periods. This ensures that callers will not abandon their call before speaking with an agent.

Furthermore, some ACD systems even offer a service to identify VIP customers and instantly route them to the appropriate agents. All of this makes it possible for businesses to quickly respond to calls and better manage your call handling process.

3. Better Agent Productivity

By routing calls effectively, your agents are better equipped to handle incoming calls. They won’t be overburdened or under-burdened as calls will be distributed equally. Additionally, less time will be spent on transferring callers to the right department or figuring out how to help a customer beyond one’s training or experience.

4. Increased Cost Savings

One of the most attractive ACD benefits is the system’s cost-effectiveness. An ACD transfers calls automatically, reducing the need for a receptionist or for employees to transfer calls back and forth. By doing this instantly, the ACD system makes it possible for the right agent to answer the call quickly, increasing first call resolution rates. Your business can improve customer service this way as callers don’t need to wait to be transferred to the right agent who knows their history or language, and so on.

Furthermore, you can subscribe to a cloud-based ACD system which is hosted by the provider. This reduces costs that would have been spent on installation and maintenance. All your business does is use the ACD service and improve call management.

5. Benefits of Cloud-Based Phone Systems

As mentioned above, cloud-based phone numbers do not need installation or purchase of new hardware. Additionally, you do not need an experienced IT team to maintain, manage, or update this hardware or software. All your business needs is a high-speed internet connection and you can use the service without interruption.

By going virtual, your business can connect agents and employees from any location through your ACD system. This makes remote working possible as your agents can work from any location as long as they have an internet connection.

6. Streamlined Business Processes

ACD systems can easily be integrated with a business’ CRM, helpdesks, social media platforms, live chat, and lead generation tools. By doing so, agents can get a wholesome understanding of each customer’s needs, preferences, and their previous interactions with the business. Through voice over IP integration, all customer information can be viewed and tracked in one interface instead of bouncing between multiple apps and software. As such, ACD systems can streamline business processes and make it easier for agents to perform efficiently in their jobs.

7. Increased Office Efficiency

All of the above automatic call distribution benefits indicate that such an automated system can vastly improve office efficiency and agent productivity. By having calls automatically routed to the right destination, no time is wasted on providing customers with assistance and support. And by creating a comfortable and integrated workplace, employees can stay up to date in regards to their callers and collaborate better with their fellow teammates.

Using ACD in Your Business

ACD systems can greatly impact the way your business interacts with its customers and enhance caller experience. Customers do not have to wait in long lines or deal with being transferred from one person to another. You can even use interactive voice response — a component of ACD systems — to have your phone system interact with callers and provide them with multiple options and assistance. To learn more about ACD and IVR systems, speak with our cloud communication specialists at 1 (877) 898 8646 today.

SIP Response Codes: A Complete Guide

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Learn about SIP response codes, how they function, and the different types of response codes available. Understanding SIP codes can help you identify issues within your communication system.

What are SIP Response Codes?

Session Initiation Protocol (SIP) is a signaling protocol used to facilitate and control communication sessions. As such, SIP lets users make and receive calls over the internet instead of traditional phone lines. This paves way for unified communications by enabling the transmission and sharing of voice, video, and other files.

A SIP session is based on a request/response transaction. Therefore, each session consists of a SIP request and at least one SIP response. Response codes indicate the status of the SIP request when making a connection between two or more parties.

How Do SIP Response Codes Work?

SIP responses use a 3-digit response code to outline or detail the status of a SIP request. For example, was the SIP request accepted, was it a bad request, and so on. These codes are divided into 6 broad categories, namely:

  1. Informational/Provisional
  2. Success
  3. Redirection
  4. Client error/Request failures
  5. Server error
  6. Global failure/error

These codes also contain a “reason phrase” which can be varied to provide additional information or in a different language.

Different Types of SIP Response Codes

So, what are the different types of SIP response codes and what do they indicate? Important abbreviations to be aware of:

  • User Agent Client (UAC) – initiates the requests
  • User Agent Server (UAS) – responds to the requests
  • Uniform Resource Identifier (URI) – a string of characters that unambiguously identify a particular resource

Here we will look at each response code in each category in detail:

1xx = Informational SIP Responses

1xx SIP response codes are sent at any time when a connection between two parties is being created. Common 1xx codes are:

100 – Trying: The request was received and an extended search or unspecified action is being performed.

180 – Ringing: The user agent has received an INVITE (SIP request code) and is alerting the user.

181 – Call is Being Forwarded: The call is being forwarded to another destination, receiver, endpoint.

182 – Queued: Indicates that the destination is temporarily unavailable and the server has placed the call in queue.

183 – Session Progress: Provides information about the progress of the call.

199 – Early Dialog Terminated: Indicates that an early dialogue has been terminated. Usually sent by the User Agent Server.

2xx = Success Responses

2xx codes indicate that the SIP request was received, understood, and accepted. Common 2xx codes are:

200 – OK: Indicates that the request was successful.

202 – Accepted: Indicates that UAS has received and accepted the request, but it has not been authorized or processed by the server yet.

204 – No Notification: Indicates that the request was successful. However, no response will be received.

3xx = Redirection Responses

3xx response codes inform the UAC about redirections and further action is needed to complete the request or reach the UAS.

300 – Multiple Choices: The request address returned several choices with different locations. The UA can select one of several options of endpoints to redirect the request.

301 – Moved Permanently: The user is no longer at the address used in the request. The original request URI is no longer valid. A new address will be provided in the Contact header field. This address should be saved and used in the future.

302 – Moved Temporarily: A new address will be provided in the Contact header field. The UAC should try the new address. This address should not be saved for the future.

305 – Use Proxy: To access the destination and address, a proxy is required. The proxy will be displayed in the Contact field.

380 – Alternative Service: The call failed, but alternatives are noted in the message body.

4xx = Request Failures/Client Error

4xx response codes indicate that the message was not processed due to an error. The request may include bad syntax and therefore cannot be fulfilled at this server

400 – Bad Request: Indicates that the request could not be understood.

401 – Not Authorized/Unauthorized: Indicates that the request requires user authentication.

403 – Forbidden: Indicates that the server is refusing to fulfill the request, even though it has understood it.

404 – Not Found: The user does not exist in that particular domain.

405 – Method Not Allowed: The method specified in the Request-Line is understood, however, it is not allowed.

406 – Not Acceptable: The resource can only generate responses with unacceptable content.

407 – Proxy Authentication Required: Similar to the 401 code, the request requires user authentication.

408 – Request Timeout: The server couldn’t find the user within a suitable time frame.

409 – Conflict: User already registered (deprecated).

410 – Gone: The user is not available here anymore.

411 – Length Required: The server needs a valid content length before accepting the request.

412 – Conditional Request Failed: The given precondition has not been met.

413 – Request Entity Too Large: Indicates that the request message body is too large.

414 – Request URI Too Long: The server refuses to accept the request. This is because the request URI is longer than the server can interpret or understand.

415 – Unsupported Media Type: Requested message body is in a format that is not supported by the server.

416 – Unsupported URI Scheme: The request URI is unknown to the server or not supported by the server.

417 – Unknown Resource-Priority: Indicates that a resource-priority option tag was present, but without a Resource-Priority header.

420 – Bad Extension: Bad SIP Extension was used. The SIP extension is not understood by the server.

421 – Extension Required: The server requires a specific SIP extension that is not listed in the supported header.

422 – Session Interval Too Small: The request contains a Session-Expires header field with a duration or interval that is too small or below the minimum.

423 – Interval Too Brief: Similar to 422, the expiration time of the resource is too short.

424 – Bad Location Information: The request’s location content was unsatisfactory or “bad.”

428 – Use Identity Header: An Identity header field is required by the server policy and one has not been provided.

429 – Provide Referrer Identity: The server has not received a valid Referred-By token on the request.

430 – Flow Failed: A specific “flow” that was sent to a user agent has failed. However, other flows may succeed.

433 – Anonymity Disallowed: The request was rejected since it was anonymous.

436 – Bad Identity Info: The request has an Identity-Info header filed and the URI contained cannot be identified.

437 – Unsupported Certificate: The server could not validate a certificate for the domain that signed or sent out the request.

438 – Invalid Identity Header: Server obtained a valid certificate used to sign a request. However, the server could not verify the signature.

439 – First Hop Lacks Outbound Support: The first outbound proxy doesn’t support the “outbound” feature.

440 – Max-Breadth Exceeded: A client that received a 440 response can interpret that its request did not reach all possible destinations.

469 – Bad Info Package: A 469 response indicates that the receiver is not willing to accept this Info Package.

470 – Consent Needed: The source of the request did not have the recipient’s permission to make such a request.

480 – Temporarily Unavailable: The recipient is currently unavailable.

481 – Call/Transaction Does Not Exist: The server received a request that does not match any dialogue or transaction.

482 – Loop Detected: Server has detected a loop.

483 – Too Many Hops: Max-Forwards header has reached the value ‘0.’

484 – Address Incomplete: The requested URI is incomplete.

485 – Ambiguous: The request-URI is ambiguous.

486 – Busy Here: The recipient is busy.

487 – Request Terminated: Request has terminated or canceled.

488 – Not Acceptable Here: Parts of the session description of the request URI are not acceptable.

489 – Bad Event: The server could not understand an event package specified in an Event header field.

491 – Request Pending: Server has some pending requests from the same dialogue.

493 – Undecipherable: The request contains an encrypted MIME body, which the recipient can not decrypt.

494 – Security Agreement Required: The server has received a request that needs a negotiated security agreement.

5xx = Server Errors

5xx response codes indicate that there’s an issue with the server and it has, therefore, failed to fulfill a valid request.

500 – Server Internal Error: The request could not be fulfilled due to some unexpected condition.

501 – Not Implemented: The SIP request method is not implemented here.

502 – Bad Gateway: An invalid response was received from a downstream server while trying to fulfill a request.

503 – Service Unavailable: The server is in maintenance or temporarily overloaded. Therefore, cannot process the request.

504 – Server Time-out: The server tried to access another server while attempting to process a request. However, there was no timely response.

505 – Version Not Supported: The SIP protocol version in the request is not supported by the server.

513 – Message Too Large: The length of the request message is longer than the server can process.

555 – Push Notification Service Not Supported: The server does not support the push notification specified in the SIP URI parameter.

580 – Precondition Failure: The server is unable or unwilling to meet the constraints specified in the request.

6xx = Global Failures/ Global Error

The request cannot be completed at any server.

600 – Busy Everywhere: All possible destinations are busy.

603 – Decline: Destination cannot participate in the call and there are no alternative destinations.

604 – Does Not Exist Anywhere: The requested user does not exist anywhere.

606 – Not Acceptable: The user’s agent was contacted successfully. However, certain aspects of the session description are not acceptable.

607 – Unwanted: The call is unwanted by the recipient. Future attempts are likely to be similarly rejected.

Buy Quality SIP Trunks from United World Telecom

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Related: SIP Trunk Pricing Breakdown (2020)

IVR versus ACD: What is the Difference?

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Both interactive voice response (IVR) and automatic call distribution (ACD) can help your business deal with high call volume more effectively. Both automated systems can direct callers to the right employee and also ensure a proper distribution of calls. You must understand the differences between IVR and ACD in order to choose the right solution for your business.

Read on for a detailed comparison of IVR and ACD.

IVR Versus ACD: Definitions

Interactive voice response and automatic call distribution are telephony automation tools that facilitate efficient call management. Both IVR and ACD can answer calls, distribute calls, and assist the callers. Both systems also provide assistance to employees and agents to help boost productivity.

Let’s discuss how these systems work and how they differ from each other.

How Does IVR Work?

Many businesses use IVR to streamline their business calls and facilitate effective call management. Interactive voice response is a voice menu that automatically answers incoming calls and assists callers. More specifically, an IVR welcomes the caller and offers menu options to identify the purpose of their call. For example, Welcome to [company name], Press 1 for Customer Support, Press 2 for Sales, and so on.

Callers select the option by either entering a number through the dial pad. Then, the IVR directs them to another set of options or transfers them to the right department or agent. In fact, some advanced IVR menus may even allow callers to complete predetermined actions such as:

  • Activate a service or account
  • Process payments
  • Send callers to voicemail
  • Record a complaint
  • Provide company and product info

The IVR system ensures that callers reach the right department or can resolve issues by themselves. In fact, some callers may not even need to interact with an agent or employee. This frees employees up to work on more complicated issues and concerns. It can also help businesses save money on hiring staff as customers can resolve most issues on their own through the voice response system. Lastly, advanced IVR systems can even record and deliver real-time stats needed for tracking and studying important KPIs.

What is ACD?

ACD works similarly to IVR. However, automatic call distribution routes calls to the right agent or department based on pre-determined rules. These rules can be based on a variety of parameters such as area code or location of the call, the time the call comes in, skills required, and so on.

An ACD distributes calls based on rules input by the account manager. These rules and conditions determine how the calls will be routed. Some common routing strategies include:

  • Round robin: Distributes calls equally among agents so no one is over- or under-burdened.
  • Least-occupied agent: Sends calls to the least-occupied agent to ensure everyone is putting in the same amount of work.
  • Simultaneous ring: Routes incoming calls to simultaneously ring multiple phone numbers within a hunt group to ensure no call goes unanswered.
  • Programmed distribution: Routes calls based on specific rules such as location of caller, time of call, customer-agent history, language skills, etc.

IVR versus ACD: Difference

The terms IVR and ACD have often been used interchangeably as if they are the same systems. However, they are not. More specifically, IVR is a part of ACD and can conduct a variety of tasks within the system. So, how do they differ?

Interactive Voice Response Automatic Call Distribution
This technology allows users to receive information from the phone system. Users need to input preferences. This technology automatically routes calls to employees, agents, or departments based on predetermined rules.
Upon receiving a call, the IVR provides the caller with options and menus. Upon receiving a call, the ACD system uses the Dialed Number (DNIS) system to check the rules for processing the call.
This system can perform a variety of application functions such as activating services, customer info look-up, etc. The IVR system works within the ACD system.
Interaction with employees is not needed. If interaction with an employee is needed, then the ACD kicks in to transfer the call to the right employee.
Offered by phone service providers as well as virtual phone service providers. Offered by phone service providers as well as virtual phone service providers.
Available as a hosted service as well. Available as a hosted service as well.

Which Do You Need?

More than their differences, IVR and ACD systems complement each other. Together, both systems can provide a robust business phone solution. They can help you manage calls and high call traffic while ensuring you do not lose valuable clients. After all, a well-managed phone system can help you provide prompt and efficient customer service. A cloud or virtual call center software can help you utilize both IVR and ACD within your office phone system. Call us today to learn how you can use a cloud IVR system to offer better customer service!

8 Powerful Applications Built Using WebRTC

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Want to know how powerful Web Real-Time Communications (WebRTC) can be for an app or browser client? Here are 8 great applications built using WebRTC that are currently being used by millions around the globe.

WebRTC Applications: 8 Powerful Examples

First, what is WebRTC? Web Real-Time Communication is a communication framework that is open-source for web browsers and phones. It is a free project that gives websites real-time communications capabilities, making audio and video communication possible. WebRTC applications can be accessed through most web browsers like Chrome, Mozilla, Safari, Microsoft Edge, etc. Additionally, they can be accessed on Android, Samsung, and iOS devices.

Let’s look at 8 powerful applications built using WebRTC and how they work.

1. Google Hangouts, Google Meet, Google Duo

Since 2011, Google has been using Web Real-Time Communications and has developed multiple communication apps for personal and business use. These apps include: Google Hangouts, Google Meet, and Google Duo.

Google Hangouts was the first to offer voice and video calls as well as online messaging and SMS. Google Meet developed as an extension to Google Hangouts as a premium video conferencing tool. It supports more users as well as speech-to-text transcription. Google launched the video calling app, Duo, in 2016 for Android and iOS users. Its use of WebRTC has led to peer-to-peer connectivity and end-to-end encryption, making it secure and reliable.

2. Facebook Messenger

Facebook’s mobile app and web client (accessible through a web browser) are both powered by WebRTC. By using Web Real-Time Communications, Messenger has brought voice and video calls to its users, and more recently, allows for co-broadcasting via Facebook Live. Additionally, Facebook has also incorporated WebRTC in VR Chat for video calls in Oculus, Workplace by Facebook, and IG Live Video Chat.

3. WhatsApp

Started as a simple messaging service, WhatsApp has grown into a global messaging platform connecting users from around the globe quickly. WhatsApp’s Android and iOS apps heavily use WebRTC as well as utilize SIP calling for fast and reliable virtual communication.

Since its inception, users can now send voice notes as well as make voice and video calls over the internet. Additionally, more recently, WhatsApp became web-accessible through its web client web.whatsapp. Users log into web browsers and use a QR code to access their messages through the browser.

4. Amazon Chime

Amazon, like the many apps and services it has offered over the years, also has a video conferencing tool called Chime. Chime is an internal video conferencing tool that uses Web Real-Time Communications in its services including Kinesis Video Streams, and Alexa’s smart home integration (cameras and doorbells). It seems that these applications have integrated WebRTC with existing communication technology such as VoIP and SIP systems.

5. Houseparty

Houseparty, the app of 2020, is a group video chat that became popular during Covid-19 lockdowns. The pandemic led to social distancing and a desperate desire for social interactions. As such, people started to look for online services that would help connect them with their loved ones. Enter: Houseparty. Using WebRTC, Houseparty provides real-time group communication and peer-to-peer video chat. Even though the rise of this company can be attributed to the pandemic, its services and popularity are here to stay.

6. GoToMeeting

GoToMeeting had used various VoIP technology and WebRTC functions in their web client video conferencing. Most of their customers and users have largely utilized the desktop client (non-WebRTC). However, growing popularity with the easy-to-use web client is drawing more users to use the browser tool.

7. Discord

Originally developed for the online gaming community, Discord combines Web Real-Time Communications and VoIP to bring voice calls and in-app messaging to its users. Discord’s engineering blog details how they have used WebRTC to serve more than two million users concurrently. They have over 87 million registered users and about 14 million active users daily.

8. Snapchat

A social media favorite, Snapchat is an app used by millions among the younger generation. Originally a platform for sharing ‘snaps’ of everyday life, the app now also boasts a video chat feature. This feature comes after Snapchat acquired AddLive, a WebRTC company that provided voice and video chat to the app.

What Can You Do with WebRTC?

As you can see, companies have used Web Real-Time Communication to develop stronger apps and browser clients. And as a result, they have made communication across boundaries quicker and more reliable. Your business can also improve its overall communication system and provide customers with better communication with these applications. To learn more, speak with our experts today!

Understanding Voice Over IP Jitter, Latency, and Packet Loss

The key to good VoIP call quality depends on a few factors such as jitter, latency, and packet loss. We discuss these elements below so you can ensure your business has strong and reliable VoIP quality for customer calls.

Understanding VoIP Call Quality: The Basics

Voice over IP or VoIP calls occur over the internet by transmitting voice or data packets from one user to their destination. On VoIP calls, your voice is transformed from analog to digital signals in data packets and is sent to your destination. Upon arrival, these packets are converted back to analog and the audio is heard. Data packets generally contain about 20 milliseconds of audio and this whole process occurs at lightning speed.

And while this process seems simple and straightforward, there are a few factors that can affect the quality of the call, interrupting it. Voice over IP call quality depends on keeping the following elements to a minimum:

  • Jitter
  • Latency
  • Packet loss

Let’s look at these issues more closely and ways to troubleshoot them.

Voice Over IP Jitter

For a VoIP or SIP call to take place successfully, data packets must be transmitted from one user to their destination. And these data packets travel through different paths before they reach the destination. As such, all data packets may not take the same path or time to arrive.

VoIP jitter refers to the data packets being delivered to the destination at irregular intervals instead of being delivered at the same time. In other words, one packet is delivered after the rest of the packet. This can lead to low VoIP call quality with missing or jumbled audio.

How to fix this issue?

Generally speaking, 30 milliseconds (or less) jitter is acceptable. However, more than that can lead to serious call quality issues, affecting your calls and customer care efforts. And so, to fix jitter issues, you must first check your network and ensure you have a good internet connection.

Another way to fix jitter issues is by using a jitter buffer. This is a space where packets are collected and stored. Then, they are sent out at regular intervals ensuring they move in the right order.

VoIP Latency

Voice over IP latency refers to lag or delay within the call. More specifically, it’s the delayed time between a caller speaking and the receiver hearing the audio. This lag or delay can lead to speakers talking over each other or echoes in the middle of the call.

It is also important to note that international calls may experience more latency than domestic or local calls. And while it is not desirable, users generally tend to accept latency in long-distance calls more than local ones.

How to fix this issue?

Latency does not necessarily affect VoIP call quality. However, it does make the caller experience less desirable, giving way to frustration and miscommunication.

Most of the time, latency is a result of network congestion, which also contributes to jitter. To combat this, you should prioritize voice over IP data ahead of other data transmitted across your network. And a high-quality VoIP router can help with this as well as other issues that may crop up within a VoIP phone system.

Voice Over IP Packet Loss

Understanding packet loss is pretty straightforward. It refers to data packets lost during transmission from one user to their destination. Packet loss occurs when:

  • Data packets are lost and never arrive at the destination
  • Packets arrive late and are discarded as a result
  • Packets contain errors and are discarded
  • High data packet loss which results in low VoIP call quality or missing pieces of audio.

When data packets go missing, communication between two parties is incomplete or unclear. Troubleshooting this issue is similar to fixing jitter and latency: check your network. Congested networks where multiple and large files are downloaded or uploaded or transferred can lead to packet loss. Therefore, to ensure low to no packet loss:

  1. Make sure you have enough bandwidth.
  2. Minimize network congestion (don’t stream videos or download music or send large email attachments).

Get a Reliable VoIP Provider

To ensure you do not suffer through these issues, it is important to find a VoIP number provider that can handle your voice over IP traffic. Learn more about our VoIP service by speaking with one of our experts today. Call us at 1 (877) 898 8646 or chat with us online for more information!