Sip Trunking guide navigation:
SIP Trunking uses VoIP to connect a PBX and the Public Switched Telephone Network (PSTN). It is an application layer protocol that sets up audio or video sessions between two devices. As such it replaces the traditional phone trunking. SIP requires on-site PBX and a connection to your Internet telephony provider.
Trunking to a Hosted PBX is typically done using SIP. Another solution would be SIP PRI (Primary Rate Interface). When compared, SIP is the more scalable and profitable service. While PRI offers better quality of voice transfer but has some limitations to its usability.
With SIP trunks, companies can connect multiple channels to a cloud PBX. This allows companies to make simultaneous calls over the existent Internet connection. Without restrictions to the number of incoming or outgoing concurrent calls.
Why SIP Trunking:
- Costs less
- Scales easier
- Is more reliable
SIP manages signalling and access to a Voice IP network. Trunks are a mediator between your phone systems and your Internet Telephony Provider. In other words it provides a connection between your telephony system and your provider of choice.
With SIP Trunking need of a physical connection to a phone company is obsolete. You no longer need that old wiring, or the circuit boxes used for connection to the PSTN.
Main sell points of Sip:
- Cost Savings
- Disaster Recovery & Security
- Cheap Long distance calls
- Extremely Scalable
- No infrastructure
- Remote access to lines
- Less constraints
- Cost Savings
SIP trunking saves a significant sum of money reaching from 20 up to 60% of company telecommunications costs. Additionally it saves time and if implemented correctly reduces stress.
SIP trunks are virtual and provided over Internet making it easy to add or remove capacity. It’s easy to scale, add or remove team members when required, often at minimal costs and without hardware requirement.
Hardware and maintenance are almost virtual as well. No wiring is required, no on premise hardware is needed and maintenance is not mandatory. Even tech support is often remote.
In case of emergencies, Sip trunking is actually quite helpful. It is easy to setup a rerouting, and backing up with a spare internet provider is not that expensive. The service is Cloud based and does not have many of the limitations of on site PBX systems, especially when it comes to power losses. Almost every provider has a failover countermeasure and recovery strategy.
As every technology Sip trunking has its security exploits and vulnerabilities. Data safety is a big concern for most companies and should not be neglected. You can find a list of possible glitches here. Any Sip trunk provider should walk you through the possible breaches as part of the provided service.
SIP can be used with VoIP apps that come a very high degree of mobility. It allows company employees’ to connect with mobile devices or from remote computers. Some VOIP/SIP providers have even developed VoIP applications that require no data connection. Those apps can be used everywhere a mobile connection is available. With them mobile workers can remain connected to the communication system, and be reachable at all times.
SIP supports remote access which is particularly handy for companies with offices based in multiple locations or countries. For outgoing calls it allows the use of all available numbers, no matter where you are located at the moment. Thus redirecting the incoming calls to your office of choice. SIP is extremely scalable, allowing startups and fast growing companies to open new sites or use full-time remote workers regardless of geographical location.
Moving and expanding the company infrastructure before SIP trunking ware challenging. Both costed a hefty sum and ware lengthy processes. With SIP adding a new line is as simple as putting on a new pair of headphones and scaling your service agreement. Off-site workers are easily integrated with applications and can be reached with a simple extension number. Moving is fast and virtually free as no on-site hardware is required.
Traditional telephone service is often reliable. But it can still interrupt at times, because of construction works, bad weather conditions, or simple human errors. And if the line is broken there is no viable workaround. SIP trunking is a bit more reliable solution. Internet backup line is almost a must and guarantees calls will get through even if the main data provider fails. Cloud servers often have backups and can be restored within minutes, and vendors often offer mobile failover. Mobile VOIP applications can take over the SIP trunking functions until the service is restored.
Extensions management and functions like call pickup, call routing can be managed through simple administrative portals with a few clicks.
Most providers offer full control of all the outgoing and incoming calls, as well as real time monitoring and statistics. It is easy to check who used which phone with which extension, who was called and now long the calls lasted.
In addition, SIP trunking also:
- Works with most modern phones
- Can use use your old phone numbers
- Can help you rationalise your telephony
- Is perfect for seasonal companies
- Can have flat plans
Migrating all your communications to SIP always comes with the need of better bandwidth. Depending on the number of simultaneous lines that are to be used the additional cost might vary from a small sum to a small fortune. Despite the ne Internet connection costs, the final balance is still in SIP Trunkings favor.
When a company starts using SIP trunking it often chooses the wrong plan. The number of simultaneous calls is mostly ill-considered and needs to be re-assessed. Companies often miss-judge how many lines they will need and need to change their SIP and data plans.
“To determine the number of simultaneous calls you have to take into account both incoming and outgoing calls. Calls that exceed the limit will receive a busy signal and won’t wait in line for pickup.”
When switching to SIP all your DID numbers can be kept but have to ported to the new system. This process is not complicated but can take a good amount of time.
Possible SIP Trunking problems if it is not implemented correctly:
- Call recording – can become messy
- Fax – highly dependant on providers and might not be supported
- 911 – proper implementation is required
- Long carrier lead times
- Number porting can take months
- DTMF might not work
SIP trunking migration might become a complicated process if not planned and executed correctly. Here are some tips that will help the implementation:
Plan everything. Begin with inventory of the existing infrastructure. Number of phone lines, DIDs, check the overall usage both outgoing and incoming. Note if you will need to use the system from other locations. Don’t forget to include the backup system.
SIP Trunking plans and carriers are flexible but you should make some firm decisions.
In the end it is not all about the money. But the cost should not be neglected. Not all carriers list the following costs for comparison yet if asked they should provide the numbers:
- Initial Implementation
- DID porting
- Initial Training
- DID numbers
- Toll-free numbers and features
- Number of lines
- Caller Identification
- Cloud Services
When you have determined what you need from your provider, and who it will be you should start planning the implementation. Every SIP provider has its course of action and you should make sure it is lined up with your needs. Be sure to take a test flight before you start the full implementation:
- Create a test plan
- Test the quality of service
- Create and test new DIDs
- Port few of your existing DIDs for testing
- Assign few checkers form your team that will gradually and troughtly test all new changes and options
- Test the internet bandwidth and backup
- To test the provided support, check time to action and availability
- Check the toll-free and 911 numbers for traffic and availability
- Check the control panel, reports and call logs
As with every new undertaking, it is common for problems to arise. Consider several potential pitfalls areas that should not be neglected:
- Call recording, and storage might not work correctly or be messy
- Faxing might not work or need to stay on traditional line
- Tall free numbers and 911 numbers might not be connected or misbehaving
- Carrier screening or spoofing of outbound Caller IDs
- Carrier lead times might be too long
- Number porting might take too long
- Control panel might not work correctly
- Call logs reports and statuses might misbehave or be hard to read
- DTMF might not work
SIP Trunking isn’t an alternative to hosted or on-premise PBX. It’s an alternative to publicly-switched telephone network. Is well-established and can lead to significant improvement in reliability and cost savings. The offered flexibility is a big plus but as any other solution it has more than a few disadvantages.
Still SIP Trunking is the right choice for a lot of companies which have left high priced and unreliable phone lines in the past and moved their business communications to the Cloud
VOIP is a telecommunications technology that replaces and upgrades the legacy Public Switched Telephone Network cabling. Its only drawback is that it requires Internet, but in return VOIP provides secure access and lots of extras important to business telephony.
Transferring to VOIP is almost mandatory if you are running business that relies on telephony or if you are simply planning to change your telecom system. VOIP offers important benefits, both financial and operational. It simplifies the infrastructure and reduces call, maintenance and moving costs.
Growing numbers of businesses have already moved to VOIP realizing these benefits. The trend is not surprising considering the lower costs for businesses. According to analysis switching from legacy systems to SIP trunking lowers the total cost by 50%.
VOIP gives the business access to the all the benefits of Internet telephony. With the new infrastructure Voip connects your PBX to an Internet Telephony Provider, benefiting you from the lower cost of long distance calls and professional pbx functions. You can even use rented cloud PBX to lower maintenance and infrastructure costs.
If have interstate branch offices and want to connect them to the communication facilities of head office, you can connect them with VOIP. This can be done by connecting the sites to your infrastructure or using a service provider that can connect your facilities with cloud PBX. Both ways, you provide your branches with improved communications eliminating duplicate infrastructure while effectively halving the support staff needed.
Operate separate networks for data and voice is not cost effective and is even harder to backup. You can make significant savings with VOIP transition. using a single channel for data, voice and video will reduce network complexity and you will cut the costs for upgrades, backup lines and maintenance.
As mentioned before the main disadvantage of VOIP communications is the requirement for internet connection. It is not much of a problem when you are calling from the office but field workers don’t have that privilege. Some VOIP providers like MIXvoip recently solved that issue by using DTMF allowing mobile VOIP calls to be made even without internet connection.
With that many benefits, how would you proceed when choosing a VOIP provider? Which are the most important factors to consider?
As people that have been in the VOIP business for quite some time now, we recommend you to choose by:
Security the most important factor because VOIP is an Internet technology. As such it can be subject attacks and other risks that come with that environment. Service providers offer different security measures and techs. Software measures include authentication, encryption and the use of secure real time transport protocols. Hardware security is mostly done with the use of separate network access routers it is called transport-layer security and is used to protect your network against attacks that can disrupt the phone service.
Quality of support really important too and definitely should be considered. As with every service, at some point problems will arise and you should be confident that your provider can minimize the downtimes. VOIP is an Internet service often hosted on cloud servers, as such businesses don’t need support staff on site, the service provider support teams have the responsibility to keep everything running.
Pricing and mobility are mostly self explanatory. They are both very business specific and depend on individual business structure and needs. Some providers charge flat, based on number of devices, others charge based on the number of calls your business makes. Most VOIP providers do not offer unlimited calls but there are exceptions. If you have lots of field workers you should make sure the company offers a mobile app.
Call quality is not that important but shouldn’t be neglected. Most providers promote on call quality but you should know that sound quality requires more data. When more data is used you can make less simultaneous calls without call interferences and you will need to upgrade your internet plan.
VOIP is essential if you want to enjoy the advantages of business communications, save money and achieve business, financial and operational benefits.
The term VoIP represents Voice communication over/via IP-network. There are many advantages of having voice communication over IP network, termed as PS ( packet switching), over traditionally voice communication, over CS( Circuit switching), like fast, easy and efficient resource utilization with enhanced features. People who wish to communicate with each other with VoIP must support TCP/IP packing, called IP-nodes/Host. VoIP is also used to establish sessions/calls to different networks like circuit switches PSTN or GSM/UMTS users.
It is a technique of sending voice/media packets over IP-network. This solution of establishing session between two parties is adopted by next generation mobile communication like 4G. VoIP is incorporated with underline protocol stack like SIP, H.323, SDP, TCP/IP suit, etc. VoIP solutions are mainly driven with the help of supporting APPLICATION layer protocol, namely SIP (Session Initiation protocol) or H.323.
Predominantly SIP is a versatile text based protocol mostly used in enhance technology. SIP defines request and response mechanism like INVITE, Register, Option, Publish etc and 1xx, 2xx, 3xx, 4xx, 5xx, 6xx responses respectively. It also defines routing mechanisms with the help of SIP-proxy server to discover called user details. A session is established between two nodes (IP-based) who wish to communicate by routing request to proxy server and querying database termed location server. Once two nodes (calling party and called party) learn each other IP-addresses, voice RTP packets are sent between them. SIP stacks maintains and control the established the voip session until it is closed. Session is released or cleared by sending BYE request to other party. SIP protocol standard is defined by IETF standards of 3261rfc and is being adopted by many new upcoming technologies. H.323 protocol on other hand is a protocol suit used earlier for VoIP solution. It is still used within certain old technology and enterprise solutions.
This section describes VoIP architecture and solution in details based on SIP protocol. VoIP architecture is defined based on three tier model.
• User-plan or Application layer mainly consists of IP-nodes who are either Calling User/Nodes or Called User/Nodes. SIP based IP-nodes are termed as UA (User Agent). SIP based IP-Node that initiates session request is termed as UAC (User Agent Client) and SIP based IP-nodes who respond to the request is termed as UAS (User Agent Server).
• Middle tier is a Server-plan or a Routing plane which help in routing SIP based request to the destination and fetching relevant data. SIP based server like SIP-proxy server, Promedia Server, Gatekeeper (for H.323 protocol) is grouped in this tier. Promedia servers are used for establishing session between SIP and H.323 based UAs.
• Database tier of the VoIP architecture consists of servers who maintains data regarding User agents, session billing and session authentication and other. Location server, Registrar server, Authentication server, Billing Server, IVR server are such example servers of this layer. The protocol which is used between the SIP proxy server and the database layer server is RADIUS, which is mainly transaction based protocol.
SIP session/signalling are done using either TCP or UDP as a transport layer, with TCP as a recommendation to achieve fast session establishment.
• SIP based IP-node which is either a calling party or called party, is configured with IP-address and can be located anywhere physically and connected through internet.
• Called Party must be registered to the VoIP network via preconfigured REGISTRAR Server IP-address and Proxy server
• Calling Party has to be VoIP registered in order to initiate a call to the called user, it must also be configured with a Proxy server
• Called Party registers itself by sending REGISTER request to the Proxy Server. The Proxy server in turn updates the REGISTRAR server with Called Party IP-address, and/or domain and/or MSIDN/sip-URI. SIP-URI is nothing but a Unique Resource Identifier in sip format like ‘sip:[email protected]’. This association created in REGISTRAR server is uploaded into Location Server which is called as AOR – Address of Record. In most cases both REGISTRAR server and LOCATION server can be colocated into one box.
• Calling Party that wish to establish a communication with called user, sends INVITE request to the Proxy or Out-bound Proxy server with called party MSISDN or SIP-URI. Outbound Proxy server is nothing but a proxy server, configured separately for handling calls/session, which is again a deployment specific. Here Calling User will act as a UAC- User agent client and Proxy server acts as UAS – User agent server.
• Out-bound Proxy server receives INVITE request, response with 100 Trying response to the Calling user/UAC acknowledging Calling user that INVITE request is received and is being processed so it does not send multiple INIVITE request. The proxy server fetches called user AOR from the location server and receives the called user IP-address.
• Proxy server routes the INVITE request to the called user IP-address received in AOR from the location server.
• Called Party upon receiving INVITE request, will to start ring with tone, notifying the user about the incoming call, acknowledging back with 180 ringing response. In the 180 Ringing, called party adds its own IP-address which is traversed backed to the calling party.
• When called party picks the calls, it response with 200 OK.
• Calling party upon receiving 200 OK response send another SIP request ACK, which indicates the flow of voice RTP packet.
• Session can be broken by either party, by sending a BYE request.
Voice packets are being transmitted between calling party and called party directly end to end in particular codec format. SDP – session description protocol suit is also deployed along with SIP stack which maintains all voice packets stream and their measurement reports. Both RTP and RTCP packets are sent over UDP as a transport protocol to achieve real time streaming. RTCP provides feedback about RTP packets sends in reverse direction.
SIP based protocol leverage to add call features like call hold, Call Park, call conference, both audio and video call, multi user support and many other call functionality. It is easy to deploy and flexible enough to twist to achieve desired result of Voice over IP.
VoIP solution is key to fourth generation mobile communication supporting multimedia services like audio, video, chat, conference, announcement services. This technology is commonly known as IMS – IP Multimedia Subsystem, a standards of 3GPP project. SIP session createс high voice quality via dedicated bearer reserved for each SIP session. If the access type is LTE, then the voip session is termed as VoLTE( Voice over LTE) and if access type is Wifi, the voip session is termed as VoWIFI.