Tag Archives: sip trunk voip

SIP Trunking Features VoIP Resellers Need

Enjoy V1 VoIP SIP Trunking solutions for resellers with features for your business voice video data and mobile communication solutions

When it comes to reseller customers, V1 VoIP understands that no to clients are alike. That’s why we carry the latest SIP Trunk features that resellers need to provide their customer incredible cost-saving services in an organized, unifying fashion.

With SIP Trunking, resellers can unify their business customer’s voice, video, data, and mobile communication systems together to create trusted, reliable, and flexible business communication solutions that deliver dramatic leaps in productivity and profitability. All of these systems work together in harmony with V1 VoIP SIP trunking.

Feature Description


BTN

Single 10-digit DID for inbound and outbound call placement. This includes E911 services, a single directory listing and a single outbound caller name setup. The default CNAM configuration is the company name.


Call Logs

V1 VoIP call logs are available on the customer’s monthly invoice or by request.


Call Waiting

Notifies call recipient of a second call while a call is already in progress. Allows switching between calls.


Directory Assistance

Local Directory Assistance.


Directory Listings

Includes the registration of the BTN with the National Registry Database. Additional directory listings are available as optional add-ons.


Enhanced 911 (E911)

V1 VoIP provides 911 routing to the appropriate local emergency dispatch center.


Inbound Calling

Inbound calling allows for receiving calls from the PSTN or other V1 VoIP users.


Inbound Caller Number

Caller Name will be sent from the V1 VoIP network to the called number as requested.


Local Outbound

Outbound local calling includes all outbound calls local to the caller as defined by the LCAD database for V1 VoIP. 7-digit dialing supported.


Local Number Portability

LNP service is available for porting numbers from another provider to V1 VoIP within the same local rate center.


Long Distance (Domestic & International)

Long-distance calls may be made to any destination in the world outside of the customer’s local calling area. Unlimited Domestic LD SIP Trunks and Minute Bundles are available that include unlimited Domestic Long Distance Calling within the continental United States, Hawaii, Puerto Rico, and Canada.


Outbound Caller Name

V1 VoIP includes the setup of one caller name associated with the BTN. Additional caller name setup available as optional add-ons.


Outbound Caller Number

Caller Number will be sent from the V1 VoIP network to the called number as provided by the customer equipment.

Contact V1 VoIP today to get started with your SIP trunk solutions.

By checking this box, I agree to V1 VoIP's Terms and Conditions


Which Codec Should You Use for SIP Trunks

V1 VoIP thinks of voice quality on quite a different scale for VoIP users than with a standard cell phone call. While comparing the two might be simple, the better method when evaluating the quality of VoIP call quality is the codec. Assigned and a scale called the mean opinion score.

A codec, which stands for coder-decoder, converts an audio signal (your voice) into compressed digital form for transmission (VoIP) and then back into an uncompressed audio signal for replay. It’s the essence of VoIP. Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc. Each service, program, phone, gateway, etc., typically supports several different codecs, and when talking to each other, negotiate which codec they will use.

Some codecs are royalty free, while others require licensing, often included in the gateways. Others drive the wholesale movement of voice traffic among the carriers and are used for specialty applications. Each contaisn variations within their own specification.

Common VoIP Codec Protocols
G.729 is the most commonly used codec in VoIP calling is a codec that has low bandwidth requirements but provides good audio quality.

G.711 with only a 1:2 compression and a 64K bitrate for each direction (128K plus some overhead), it is best used where there is a lot of bandwidth available.

G.722 is a high bit rate (48/56/64Kbps) ITU standard codec which, because it is of even better quality of the traditional public switched telephone network (PSTN), it can be used for a variety of higher quality speech applications. But be warned, this standard also requires an adequate amount of bandwidth.

You can assign a different codec to individual phones. You can use medium quality/low bandwidth G.729 codec while the boss and legal department uses the superior quality/heavy bandwidth G.722 codec. The codecs that provide the best quality consume the most data bandwidth, thus there is a trade-off that you need to consider. The easiest way is to ascertain, on a phone by phone basis, whether you want the voice conversation to be:

Do you need some guidance determining which codec to use for your SIP trunks? Contact V1 VoIP right now and we will point you in the right direction. N

By checking this box, I agree to V1 VoIP's Terms and Conditions


What is SIP Trunking for Resellers

The phrase “SIP Trunking” or “SIP Trunks”, is increasingly common in the world of telecom. In the last few years, the telecommunications industry has standardized on SIP as the protocol of choice for phone calls on the data network. SIP (Session Initiation Protocol) is the technology used for establishing a voice communication session on a data network like the Internet. A SIP “session” might be a regular VoIP phone call between two participants or a multi-party conference call.

What Is SIP Trunking?
A SIP Trunk provides the same service you get from a traditional analog phone line except a SIP Trunk is a “virtual” phone line which is provided by V1 VoIP. We use your data circuit (T1, Cable Modem, DSL, etc) to connect your phone system back to their network.

SIP Trunks Save Customers Money!
Because there are no physical lines that need to be maintained with a SIP Trunk, they are substantially less expensive than traditional telephone service. SIP Trunking can save you up to 50% and more on your monthly bill, depending upon the types and quantities of calls you are making.

What SIP Trunks Do You Need?
The number of SIP Trunks is determined by a few factors. Your Internet connection is an important consideration. SIP Trunks consume bandwidth. Depending upon the quantity of phone calls and need for data bandwidth, you may have to expand your data circuit. First, how many concurrent calls does your business need? As a typical rule of thumb, you’ll need approximately one SIP trunk for each 2-3 users.

V1 VoIP is a leading SIP trunk provider and we have the added advantage of providing our enhanced PBX to go alongside SIP Trunk services and solutions. Contact us and we’ll be glad to see just how much money our resellers make and how you can save your customers money.

By checking this box, I agree to V1 VoIP's Terms and Conditions


Basic Guide to SIP Trunk for V1 VoIP Resellers

A basic guide of SIP Trunks and why they are popular for V1 VoIP private label resellers

For V1 VoIP resellers, the ability to resell SIP Trunks to connect to the traditional PSTN network is a great service to add to your solution portfolio. SIP Trunks are becoming a hot selling item because of the cost savings of SIP for customers, along with it’s increased reliability.

Reselling SIP trunks allows your customers to replace these traditional fixed PSTN lines with PSTN connectivity via V1 VoIP SIP Trunking Service. This service allows those businesses that have a PBX installed to use VoIP outside the enterprise network by using the same connection as the Internet connection.

A SIP Trunk is primarily a concurrent call routed over V1’s IP backbone using VoIP technology. SIP Trunks are used in conjunction with an IP-PBX and are thought of as replacements for traditional PRI or analog circuits. SIP stands for Session Initiation Protocol and is widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. IT allows for video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games.

Have you waited long enough to begin reselling SIP trunks and SIP trunk service? Don’t wait any longer. Contact a member of the V1 SIP team today.

By checking this box, I agree to V1 VoIP's Terms and Conditions