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What is a SIP Proxy and How Does a SIP Server Work?

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We write a lot about SIP. That’s because we have been a leading SIP trunk provider for over twenty years. Read on to learn more about SIP servers and how they work.

What is a SIP Server?

A SIP server or SIP proxy processes session initiation protocol (SIP) requests. This server is the main element of an IP private branch exchange. SIP is an internet protocol used to initiate and receive voice and video communication by transmitting data packets across an internet connection. This enables the quick and easy transmission of SIP calling between 2 or more parties.

How Does a SIP Server Work?

A SIP server works alongside a voice over IP or VoIP phone system. Both systems together make cloud communications possible. A SIP proxy can:

  • Set up a session between 2 or more endpoints; such as audio or video conferencing between 2 or more parties
  • Replace one endpoint for another; during call transfer or routing
  • Negotiate and adjust media parameters and specs during a session; such as putting a call on hold
  • Terminating a session

It is important to note that the SIP server does not actually transmit media. Media transmission is performed by a media server using the RTP protocol. Within an IP-PBX, the SIP server and media server are present on the same machine. However, a high-volume SIP server like a VoIP provider may separate the two servers on different machines and balance the load.

Additionally, there is no fee or charge to get a SIP address for your server. These addresses connect to unique phone numbers. This enables each user on a SIP network to have a direct inward dialing number to place calls. Furthermore, companies can use these systems in a package such as a hosted PBX.

SIP Proxies: Modes of Operation

A SIP server generally operates in one of two modes: Stateless or Stateful.

1. Stateless SIP Proxy: This type of SIP proxy receives and transmits messages but does not keep any record of the transmission. A stateless SIP proxy works this way: Send > Receive > Delete. This server works at a faster speed because of its limited functionality. Additionally, this simplicity in functioning makes it desirable to small businesses as they can easily scale and upgrade their SIP system.

2. Stateful SIP Proxy: This type of SIP server transmits as well as stores messages and information to access later. Because of this functionality, it can pick up a request message and try again. Or, it can reroute the message through another aspect of the network. A stateful SIP proxy works this way: Send > Receive > Save. An example of this is Time of Day Routing that routes incoming calls based on the time of day and predetermined rules. For example, calls made to a business after hours can be forwarded to a different office location or remote agent.

What Does SIP Trunking Do?

SIP trunking is a service that enables your PBX system to send VoIP and SIP calls over the internet. This service works with virtual telephone lines and sends and receives messages through bandwidth data. You can get multiple SIP trunks and cover various geographic areas. SIP trunking makes it possible for your business to expand operations beyond your immediate location.

An image of a SIP proxy and server.

Benefits of a SIP Proxy and SIP Trunking

SIP servers and SIP trunking have become increasingly popular with businesses of every size. Here are the top benefits of switching to SIP:

1) Enable Unified Communications
SIP trunking enables voice, video, and text messaging from one platform. There’s no need to invest in different services to keep your communications stable. You can make and receive high-quality calls, audio and video conferencing, and texting from SIP trunking.

2) Forward Calls
With a SIP server, you can quickly and easily forward or direct incoming calls to several SIP and VoIP devices. This is helpful for any office with a busy call volume. Plus, with a stateful SIP proxy, you can save and access calls or messages that didn’t go through the first time and try again later.

3) Cut Communication Costs
VoIP and SIP are in demand because it not only offers high-quality communication but also a comfortable price. Sending and receiving voice, video, and text over a SIP server costs almost nothing to users.

4) Network Security
Secure VoIP is a necessity within any business. A SIP server protects your communication system from hackers by disconnecting calls and users without credit or authorization.

5) Access to VoIP Features
With SIP trunking, you can gain access to useful voice over IP features that can help organize calls and provide a professional image to your business. Features include, but are not limited to;

  • Call forwarding and routing options
  • Automated greetings
  • Analytics and reports
  • Unlimited extension, and so many more

Where Can I Get SIP Trunking?

United World Telecom can help! You can buy SIP trunks directly from our website or contact one of our experts to learn more.

What is SIP Calling?

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Communicating effectively with customers is crucial to the successful running of any business. It is through your customers that you can improve your product and increase sales. And so, if your business isn’t doing everything it can to make it easy for customers to connect with you, then you are falling behind. Learn what SIP calling is, how it works, and how businesses can benefit from using a SIP phone system for business communication.

SIP Calling: The What, How, & Why

To understand SIP calling (or SIP trunking), you must first be familiar with Session Initiation Protocol (SIP). SIP is a signaling protocol that initiates real-time voice calling between two or more parties. This IP is in charge of starting, maintaining, and terminating the call over an internet network. By doing so, SIP technology makes it possible for users to make and receive high-quality calls over a virtual network. Let’s look at how SIP calling works and how businesses can use SIP for improved communication.

What are SIP Calls?

SIP calls use Session Initiation Protocol to transmit voice calls over a SIP trunk or SIP channels. In other words, SIP calls are voice calls sent over an internet protocol or internet connection.

Often used interchangeably with voice over IP or VoIP calls, the two systems are different. VoIP makes SIP calls possible. This is because SIP uses VoIP technology to transfer calls from one end to another destination over a stable internet connection.

How Does a SIP Call Work?

Traditional phone systems consist of a PBX system and phone lines connecting to a PSTN.

SIP technology removes the need for a traditional, physical connection. With SIP, you do not need to be connected to a phone company or geographical location. And, you do not need multiple phone lines for different departments. You will get a SIP trunk, instead, to run your virtual phone system. You can then establish voice communications virtually via the internet.

And what’s the end result? You can get a virtual phone system with call management and call routing features (global call forwarding, outbound calling, etc.) without physical or multiple phone lines. This system can be used from any location, connecting multiple devices.

Why Does Your Business Need SIP Calling? 5 Benefits of SIP Calls

So, how can SIP calling benefit your business? From being a cost-effective alternative to creating a unified communications platform, SIP trunking can help businesses organize their internal and external communications to connect better and increase productivity and efficiency. Let’s look at the top benefits of SIP calling for business:

1. Save on Business Calling Costs
First and foremost, the cost of SIP calling is highly affordable for businesses needing multiple phone lines and with various departments. Not only can you make and receive high-quality voice calls inexpensively, but you can also bypass international calling rates when offering global customer support.

2. Scale Up or Down as Needed
SIP trunks are designed to support a business’ scalability needs. This means that if you need to scale up and add more direct inward or outward dial numbers, you can do so easily. And the same goes for scaling down; that is, removing lines or pausing certain services and features. And most of these actions can be done by you, reducing the number of times you will panic-call your SIP provider.

3. Improve Communications with Better User Experience
SIP trunks, and SIP technology generally, are easy to use. You do not need to employ new IT teams or conduct rigorous training or worry about setting up complicated software. You and your employees can simply manage everything from a user interface provided by your SIP provider. Use this interface or control panel to set up features and service, make changes or adjustments, add lines, and more.

4. Experience High-Quality Voice Calls
When using SIP, your business no longer relies on physical landlines. This means that your communication system does not easily fall apart due to power outages or weather conditions. SIP calling utilizes redundancy to automatically reroute calls from one location to another if the previous location’s user is unavailable or inactive. This is a reliable way to make and receive high-quality business calls.

5. Offer Excellent Customer Service
Never miss calls by routing them to different locations, in case the first is unavailable. Reduce the number of dropped calls or low-quality calls. All of these factors come together to help your business offer excellent and uninterrupted customer service to not only local but global customers as well.

Get SIP Calling with United World Telecom

With a SIP phone system in place, your business is gearing up to communicate with your customers conveniently and cost-effectively. Learn more about how SIP trunking can boost business communication by chatting with our experts or calling us at 1 (877) 898 8646.

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)

SIP Line vs. SIP Trunk: What’s the Difference?

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Many businesses offering local and international services often wonder if there is any way to modernize and enhance communications without receiving overwhelming bills each month. Well, SIP trunking and SIP lines make a fine solution. Let’s look at the difference between a SIP trunk and line so you can get the phone system that works best for you.

SIP Line vs SIP Trunk: Understanding the Difference

Oftentimes the terms SIP line and SIP trunk are used interchangeably. However, there are two components that work together and are not the same thing. Here we explain what each term entails so you have a better understanding.

What is a SIP Line?

SIP lines are also called SIP channels or SIP sessions. A SIP line is an element of a SIP trunk through which data can be exchanged between two points. A SIP trunk holds large numbers of SIP lines or channels. During calls, a SIP channel is the unit’s capacity to support an incoming or outgoing call. In other words, each call uses one channel.

And so, companies needing SIP lines are more than satisfied by one SIP trunk. However, exactly how many lines your business needs depends on the number of concurrent or simultaneous calls you handle per day. SIP providers offer a variety of plans:

  • Unlimited channels
  • Unlimited channels with a fixed number of session minutes
  • Fixed number of channels with unlimited minutes
  • Multiple trunks for multiple offices on the same network

What is a SIP Trunk?

SIP trunks connect your business’ private branch exchange or PBX system to the internet. In other words, it converts your traditional phone system into a virtual or digital version.

As noted above, a SIP trunk holds SIP channels and makes concurrent SIP calling possible. In VoIP phone systems, a SIP trunk can hold about a hundred SIP channels. These lines can be distributed across different departments, devices, and so on.

SIP trunking enables your business to make and receive calls over the internet instead of a traditional POTS line. This way, this method of communication supports both traditional as well as VoIP phone systems. With SIP trunking, you can transmit voice, video, text, and other unified communications.

Benefits of SIP Trunking

Why should your business consider SIP trunking as part of your communication system? Here are some ways SIP trunking benefits businesses:

1. Unified Communications: With SIP trunking, you can handle voice, video, and text transmission through one platform. Additional services are not required.

2. Low Cost and Set-up: SIP trunks don’t need extensive hardware or software to function. A reliable internet connection is more than enough.

3. International Coverage: With SIP trunking, your local and international business calling bills reduce tremendously.

4. Scalability: You can always add extra SIP lines to your SIP trunk if/ when needed. Again, there is no extensive set-up or installation required.

5. Tools for Better Customer Service: By providing voice, video, and text communication, you can enhance customer support by adding video conferencing, text messaging, and file-sharing.

Does Your Business Need a SIP Line or SIP Trunk?

The decision really isn’t one or the other. This is because SIP channels work within a SIP trunk. The real question is, how many channels does your business need?

A SIP trunk can host multiple SIP lines, allowing for multiple concurrent calls to occur simultaneously. In other words, if your business has 250 SIP lines, it can handle 250 calls at the same time. Each call takes up 1 SIP line.

Similarly, SIP trunking prices or monthly fees will depend on how many lines your SIP trunk holds. To determine this, you will need to know how many incoming and outgoing calls occur during a day. Work with your sales, marketing, and customer service teams to get an approximate number.

If your business does not receive or make a large number of calls per day, then you may choose to have a few employees or small teams on each SIP line. Additional SIP lines can be added later.

Connect SIP Trunks to Your Phone System Today!

Think SIP trunking is the way to go? We at United World Telecom can help you buy SIP trunks. Add premium call management features such as call forwarding, call recording, IVR, outbound calling, and more. Call us today to sign up!

What is Session Initiation Protocol (SIP)?

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In this comprehensive post, we will explain Session Initiation Protocol (SIP) and its important features.

What is Session Initiation Protocol?

Session Initiation Protocol (SIP) is an internet signaling protocol that is used to initiate, maintain, and terminate real-time voice and video communication. An internet protocol is a set of rules for routing packets of data across a network. SIP is often used for mobile phone calling over LTE and in IP telephony systems.

It enables businesses to take their communications to the next level, by enabling not just voice-sharing but video conferencing and text messaging as well. Unified communications are made possible for an organization through SIP.

Important SIP Features

Note that SIP is a signaling protocol. In other words, it doesn’t handle the specifics of multimedia sessions. Other protocols such as the Sessions Description Protocol and Real-Time Transport Protocol handle media details and data delivery respectively.

Session Initiation Protocol initiates a call between two or more people where the SIP client establishes specifics of the request. The other party can choose to accept or reject it. This transmission is done securely through Transport Layer Security (TLS).

Additionally, Session Initiation Protocol makes it possible for users with different service providers to interact and communicate. It can boost your IP telephony call with a process called SIP trunking.

Session Initiation Protocol versus Voice over IP

SIP and VoIP are often used interchangeably and therefore, it is important to understand the difference between these two types of technology. See below:

SIP Trunking Voice over IP Tech
Signaling protocol within VoIP Family of technologies related to communications
Multimedia transmission Only voice messages transmitted
Requires only a modem VoIP devices need to be connected to a computer
Uses a peer-to-peer system to handle large amounts of data Uses a central network to organize and transmit traffic

Benefits of Session Initiation Protocol

Why do businesses benefit from SIP trunking and why should you consider this service for your communication system? It’s simple: SIP and SIP trunking can boost and enhance communications in a multitude of ways. It is not only cost-effective but provides phone solutions to improve internal and external communication. Here are some of the top benefits of SIP trunking.

A HIghly Cost-Effective Service
SIP calls are routed over the internet and by doing so, this gives you the ability to contact local and long-distance contacts for cheaper calling rates. Pricing for a SIP trunking service is pretty straightforward. You sign up for a monthly subscription and your bill remains predictable. Furthermore, most SIP providers do not require you to sign long-term contracts. Check out our detailed analysis of SIP trunking prices for your information.

Tools to Expand Internationally
Part of growing your business globally involves creating strong communication channels between your company and its customers. SIP trunking paired with virtual phone numbers is a great solution for businesses looking to expand their reach into international markets. You can reduce your calling costs while providing convenient (and even free) ways for customers to connect with you through local numbers and toll free service.

One Platform for All Communications
You can transmit voice, video, text, and more via SIP channels. This technology makes multimedia communication possible from one platform, instead of having it spread across 4-5 different services or software. This can help increase office productivity and reduce costs that can build up from different platforms.

More Session Initiation Protocol Resources

Contact us at 1 (877) 898 8646 if you want to learn more about Session Initiation Protocol and let an expert help you through the process!