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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.

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!

PRI Explained: What is a Primary Rate Interface?

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Choosing a business phone system for your company is a necessary part of creating the perfect communication system. With advancements in technology, there are many different systems available for businesses to choose from. Here we will discuss primary rate interface (PRI) and the advantages and disadvantages of this phone system.

What is PRI?

A primary rate interface or PRI is a communication system that is provider-free. This system allows businesses (users) to send and receive voice, data, and video files through a copper wire network. PRI systems or lines constitute two pairs of copper wires. This feature of primary rate interface networks provides secure data transmission. You can get two types of PRI systems:

  • Basic rate interface solutions (BRI) for personal and small business use
  • PRI for large enterprises and corporations.

Features of Primary Rate Interface

To understand how these communication systems work, it is first crucial to be aware of their features. Key features of a PRI system include:

  1. Lines are made of two pairs of copper wires connecting the provider and the user.
  2. You can have 23 B-channels on a single telephone line. And by doing so, it enables businesses to have multiple extensions and telephone numbers via one connection.
  3. Each channel has 64 kbps for data transmission.
  4. Can connect two private branch exchange or PBX systems together and can also work with an IP PBX system.

Advantages of a PRI Phone System

There are different ways a primary rate interface phone system benefits businesses. However, whether or not your business needs this system depends on what you hope to achieve through your business communication system. Let’s look at how PRI systems boost business communication:

1. Extensions and DID numbers:

Direct inward dialing refers to direct numbers assigned to individuals within a business. This means that callers from outside can dial this number and reach a contact directly. Extensions work in a similar way with an additional code attached to a number to let callers reach an individual or department directly.

With PRI, SIP trunking, or virtual phone systems, you do not need additional lines for each number or extension. For PRI, specifically, you can have up to 23 conversations happening simultaneously on one line. That means you can have up to 23 users using the system. And that is considering everyone uses it at the same time. If you need simultaneous communication, you can add more users to these existing lines and they can use it as and when needed.

2. Scalability and expansion:

As your business traffic grows and communication needs increase, you will want to scale and expand. And a primary rate interface will allow you to do that. If more users are needed, you can simply get another PRI line and add it to your existing system, giving 23 more users the ability to communicate.

PRI Drawbacks

While a primary rate interface system changed the way businesses communicated over the years, phones have come a long way since. Advancements in telecom technology have given rise to more modern and user-friendly systems.

The biggest drawback that PRI systems have is the ability to expand in bundles of 23. This means that if you have just one or two extra employees and all channels are used constantly, then you will need to buy 23 more channels for those extra employees. You will end up paying more than you need.

On the other hand, if you run a large corporation with 100-150+ employees, then you will need multiple PRI lines to work efficiently. Additionally, it gets more complicated if you need to add multiple locations or remote workers.

To combat these issues, you have a few alternatives to consider: Hosted VoIP and SIP trunking.

PRI vs Hosted VoIP vs SIP Trunking

Most businesses today have adopted a cloud VoIP or hosted VoIP solution. Hosted VoIP means that your service provider hosts your phone solution and takes care of all your software needs and maintenance. All you do is use the service. You do not have to worry about purchasing hardware and software, maintaining it with a professional IT team, and so on. This helps your business save on communication and IT-related costs.

SIP trunking is a session initiation protocol (SIP) feature that enables transmission of voice communication over a data network. SIP trunking works similarly to POTS except that the phone lines are virtual instead of standard copper lines. And your phone system connects to your provider via your internet connection. SIP trunking has often been used as an alternative to POTS and PRI systems.

PRI, unlike VoIP and SIP trunking, does not rely on internet bandwidth for transmission, and therefore does not suffer from jitter or packet loss. However, there are limitations in terms of scaling upwards, mobility, and features available.

Here’s a table to demonstrate the differences between these business phone systems:

PRI SIP trunking Hosted VoIP
1. Upfront costs Medium-High High Low
2. Maintenance costs Medium-High Medium-High Low-High
3. Connectivity Physical Virtual Virtual
4. Service quality Low; calls may experience muffled or distant quality, frequency range is limited High; good bandwidth required, low bandwidth can lead to jitter, packet loss High; good bandwidth required for VoIP, low bandwidth can lead to jitter, packet loss
5. Scalability Low High; very scalable High; scalable
6. Mobility None; no routing ability Medium; calls can be transferred to predetermined locations Very high; can be used anywhere and through any device

Choosing the Right Phone System for Your Business

The phone system that is ideal for your business purposes depends on what you want to accomplish with it and what your budget can include. Speak with our experts today to see if VoIP or SIP trunking is a good fit for you!

6 Ways to Fix VoIP Jitter

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When conducting business calls, interruptions, low call quality, or missing audio can lead to miscommunication. Part of running a professional business is ensuring that your business calls, whether for queries or support, occur smoothly without any distortion or jumbled audio. Interruptions during calls can lead to losing valuable clients. One important element that affects VoIP business calls is jitter. In this post, we explain what leads to jitter and how to fix VoIP jitter in 6 useful ways.

Why You Need to Fix VoIP Jitter

In order to fix VoIP jitter, one must understand VoIP jitter and how it affects a business’ VoIP phone system. During VoIP business calls, voice messages are transformed from analog to digital signals and stored in data packets. For VoIP calls to connect two end-points successfully, data packets need to be transmitted effectively without delay or disturbance.

While these data packets move from one end-point to the next, the packets travel through different paths and may not take the same path. However, due to a variety of reasons — such as low internet speed, a low-quality router, and so on — the data packets may not be delivered at the same time. Instead, they may arrive at irregular intervals affecting VoIP call quality. Additionally, this can lead to missing or jumbled audio. This is known as ‘VoIP jitter.’ Jitter within business calls can lead to miscommunication and frustration for users. Here are 6 reliable ways to fix VoIP jitter:

1. Invest in a Powerful Router

When purchasing a router for your VoIP phone system, do your research and find one that is powerful and can handle your VoIP needs, especially the bandwidth capacity. Carefully review the product and see if it matches your needs. Study customer reviews and testimonials and look for complaints and potential issues.

2. Utilize an Ethernet Cable

Use a high-quality ethernet cable to connect your VoIP system to your router. This way, you will have a better connection and no interference from sources out of your control that can lead to jitter, latency, packet loss, and more. Additionally, if you already have an ethernet cable but are still experiencing jitter, then perhaps it’s time to upgrade your ethernet.

3. Subscribe to High-Speed Internet

Next, as is widely known, low internet connection speeds can affect the quality of your VoIP phone system. Low internet speeds lead to jitter, latency, and more. Make sure that your business has high-speed internet connection to ensure smooth connectivity.

4. Conduct Bandwidth Tests

Besides securing a high-speed internet connection, you also want to ensure that your bandwidth is strong enough to carry the weight of your VoIP phone system. Ask your ISP to test your bandwidth and then resolve jitter issues. You may even connect with your VoIP phone service provider for help in resolving VoIP jitter issues.

5. Consider Getting a Jitter Buffer

Another way to fix VoIP jitter is by using a jitter buffer, a device that intentionally delays an incoming data packet. By delaying an incoming packet, the receiver of the call will hear the voice message clearly and with very little distortion. This is because the jitter buffer will re-group delayed data packets and then play them together, steadily. Your data packets will be stored in the right sequence and played accurately and clearly.

6. Reduce Unnecessary Bandwidth Usage

Lastly, make it a practice to reduce unnecessary bandwidth usage, especially during office hours. Teach your staff to not use large amounts of bandwidth for non-work-related activities. This includes streaming videos or content from Netflix, etc. These services use large amounts of bandwidth and can lead to jitter during VoIP calls.

Convert More Customers with VoIP for Business

A business VoIP phone system can greatly improve the way your business communicates with its customers. Additionally, getting this service from a reliable VoIP provider can help improve VoIP call quality issues such as jitter, latency, and so on. Ready to upgrade your business phone system and get VoIP? Speak with our representatives 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

Buy SIP trunks from us and improve the way your business communicates with advanced features, high voice quality, and competitive rates. Sign up on our website or speak with our specialists to learn more!

Related: SIP Trunk Pricing Breakdown (2020)

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!

What are the Different Types of Contact Centers?

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Can’t decide which of the different types of contact centers is right for your business communication needs? Here we highlight 7 types of call centers to help you understand which is ideal for your purposes.

Understanding the 7 Types of Contact Centers

There are a few different types of contact centers that exist with different focuses and purposes. This ranges from centers that have it all or centers that focus on incoming or outgoing calls, those that use specific virtual call center software to centers offering multichannel communication options, and so on. You can also outsource your communication needs to some centers. On the other hand, other types of call center software can give your business access to contact center tools to use in-house.

However, which type of call center or call center software that will work for you depends on your specific needs. So, let’s look at the 7 types of contact centers:

  • Call centers
  • Contact centers
  • Inbound centers
  • Outbound centers
  • On-premise centers
  • Cloud-based or virtual centers
  • Multichannel or omnichannel centers

Let’s look at each of these types of contact centers individually.

1. Call Centers

The terms call center and contact center are often used interchangeably; however, there are a few differences between them. A call center, for instance, is a centralized center where reps answer incoming calls from potential and current customers of various businesses. Some call centers handle only incoming or outgoing calls while others handle both, also called ‘blended’ centers. Additionally, a business can have an in-house or on-premise call center or they can outsource their needs to a company specializing in call center services.

2. Contact Centers

Contact centers are similar to call centers except that they are more multichannel or omnichannel. This means that along with receiving calls, these centers also offer email, SMS, live chat, and social media communication channels. Call centers usually stick to phone conversations only while contact centers offer more channels and modes of contact.

3. Inbound Centers

Inbound contact centers focus primarily on incoming calls. This means that they have trained agents and reps to answer calls and provide sales or customer support services. Most inbound contact centers are generally customer service-oriented. Customers generally call a business for a few reasons:

  1. To inquire about a product or service
  2. To ask for technical support
  3. To receive assistance with a purchased product or service

Usually, an IVR system answers the call and interacts with the caller by offering menu options. Then, it proceeds to help the caller via pre-recorded messages or by transferring the caller to the right department.

The goal of inbound centers is to handle customer calls and concerns quickly and efficiently. This means answering and resolving calls professionally. This helps retain more customers by increasing customer satisfaction.

4. Outbound Centers

Outbound contact centers do the opposite of inbound centers. That is, they focus primarily on outgoing calls and lead generation. These contact centers call lists of potential clients or leads in an attempt to make new sales. Outbound centers use customer relationship management (CRM) systems to keep track of contacts, leads, and calls. Some outbound centers offer additional outbound calling services such as fundraising, collecting customer feedback and surveys, outreach efforts, and more.

5. On-premise Centers

Many types of contact centers work on-premise or in-house. This means that the call center works within your office and all the hardware and software are operated and managed by your in-house IT team.

On-premise centers are known for their high level of data security and therefore tend to be more reliable and have better call quality. Additionally, you will have total control over your communication system and you can use it according to your needs. However, running your contact center on-premise means that your business will be in charge of purchasing and maintaining hardware, hiring a highly-skilled IT team, and paying other upgrade costs. All of this can lead to higher costs for your business.

And so when deciding whether you need an on-premise center, consider this: do you have the budget, infrastructure, and IT team to handle the system in-house?

6. Cloud-Based or Virtual Call Centers

Virtual call centers are an alternative to on-premise centers. They work virtually and are hosted by your provider. In other words, your provider runs and manages your call center software while you simply use the service.

A cloud-based contact center gives you less control. However, your business is not responsible for any hardware or maintenance costs, which can save substantially on expenses. Your teams can access the software from any location or device as long as they have a good internet connection. Most businesses that manage remote teams use virtual call center software to provide their teams with the right tools needed for excellent customer service.

7. Multichannel or Omnichannel Centers

Multichannel and omnichannel centers are one of the most effective types of contact centers that offer not just voice but other communication channels as well. This includes video, email, SMS, live chat, and social media engagement. One thing to note is that multichannel centers may not offer all communication channels while omnichannel centers do.

Having multichannel support can help your business reach more customers across different channels. Interested customers who do not prefer phone conversations can use other means to connect with your business, which helps you increase your customer base.

Choosing the Right Call Center Software for Your Business

Businesses of every type can use call center or contact center software to improve the way they interact with their customers. You can enhance caller experience, increase customer satisfaction, and in turn, improve your business’ overall sales. Reach out to us today to learn more!