Cloud Nippon provides complete VoIP telephony solutions to help you become a truly 21st century business. A SIP TRUNK is a direct connection between your organization and an Internet telephony service provider which is a simpler, easier and less expensive configuration. Cloud Nippon’s high quality Sip trunks are certified with major IP-PBX service providers like 3CX, Askizia, Allo, Asterisk, Grandstream, Cisco, Avaya, Beronet, Yealink, Hanlong, etc.
A SIP TRUNK is a direct connection between your organization and an Internet telephony service provider (ITSP). It enables you to extend voice over IP (VoIP) telephony beyond your organization’s firewall without the need for an IP-PSTN gateway. This simpler configuration is easier and less expensive to design, operate, maintain, and upgrade.
A SIP (Session Initiation Protocol) Trunk refers to the communications setup where your organization and an ITSP (Internet Telephony Service Provider) enjoy a direct connection. This type off a connection allows your organization to implement VoIP telephony without resistance from a firewall and without assistance from an IP-PSTN gateway.
SIP Trunk Malaysia configurations are much simpler in terms of operation, maintenance, and upgrading. Plus, they are inexpensive to design, which means your organization will not have to spend large sums of money to incorporate such advanced technology into the existing IT infrastructure.
SIP Trunking also does away with PRIs (Primary Rate Interfaces), PSTN gateways, and BRIs (Basic Rate Interfaces), which results in telephony costs being reduced drastically. Extending the system over IP is an easier process and also, inexpensive.
SIP Trunking prevents data and voice connections from being separated. It, in fact, maximizes the potential for the convergence of communication. Most importantly, SIP trunking is scalable because the infrastructure needed to accommodate growing data and voice traffic is already installed. For instance, an entire enterprise can be served with just one SIP Trunking account. This can be highly advantageous for enterprises that operate multiple sites as they do not need to rely on sub PRI connections for each site.
With Cloudnippon’s SIP Trunking service, your organization can turn every call into a local call. SIP calls are transferred via the internet or a specific IP network to a particular termination point, from where the call is forwarded to a local PSTN.
Your organization will not be requiring any of those expensive toll-free numbers to gain customers. Our SIP Trunking service is available across multiple locations and we can create local numbers for your customers in each of these locations. After local termination, the calls will be transferred via the internet to your organization’s call center.
Cloudnippon also provides Telephone Number Mapping with its Cloudnippon SIP TRUNK service. With Telephone Number Mapping, we can match caller numbers to existing SIP addresses. This will allow calls to be completed over the internet. The end result being higher savings for your organization. Even if the call cannot be completed, it will automatically be transferred to PSTN.
Investments are relatively much lower when it comes to installing an SIP Trunk service. At an enterprise level, all you would need is an IP-PBX, IP Phones and most importantly, a SIP-aware firewall for security.
Cloudnippon offers one of the best SIP Trunk services in the market. We provide state-of-the-art SIP TRUNK for IP phones and IP-PBX. WE also enjoy certifications from top IP-PBX system providers such as Cisco, Beronet, Asterisk, Avaya, 3cx, Hanlong, Askozia, and Yealink.