SIP Trunking

Are you looking for a SIP Trunk?

We provide various SIP-trunking services & solutions to cater to your individual business needs.

SIP Trunking


SIP Trunking Overview

SIP (Session Initiation Protocol) is a real-time communication protocol for VoIP.

To summarize, with SIP Trunking, the IP media stream coming from within the enterprise stays as an IP media stream passes to anywhere within the enterprise or across the boundary of the enterprise to another enterprise via IP. This reduces the need for hardware media gateways at the enterprise edge and carrier edge (often referred to as the PSTN) completely and provides significant savings to the enterprise.


Quality of Service

RTCP (Real-Time Control Protocol) packets are used to provide QoS measurement reports and other information. The VoIP RTCP-XR-eXtended Reports MRB-Metrics Report Block provides measurements (metrics) for monitoring quality of VoIP calls and conversations. These measurements include packet loss and discard metrics, delay metrics, analog metrics, and voice quality metrics.

The MRB (Metrics Report Block) reports individually on packets lost (discarded) on the IP channel as opposed to packets that have been received and then lost by the receiving jitter buffer. MRB reports on the combined effect of losses and discards which can be used to determine corrective actions on voice QoS. Network analyzers or probes gather RTCP-XR packets and provide reports for management evaluation. This report provides details on Delay, Packet Lost, Packet Discard, Jitter, R-Factor, MOS-Mean Opinion Score and other factors.


Unlimited Growth

In addition to PBX hardware savings, there are savings in regard to network (bandwidth) connections. Rather than the traditional 24 channel “stair step growth,” SIP Trunking supports virtually unlimited incremental or scalable growth sometimes referred to as N-way (un-limited) growth. In other words, think of voice not as fixed channels but as data packets which share the bandwidth with other data applications. Only as needed is bandwidth then added.

SIP Trunking also allows a customer to oversubscribe (increase capacity) the number of voice calls by utilizing advanced compression techniques such as G.711 and G.729 CODECs-COmpression-DECompression (or CODer-DECoder) devices increasing capacity by 400% or more. Check with your provider for specific features and options.


Total Cost of Ownership

SIP Trunking can also provide significant lower TCO (Total Cost of Ownership) and operational cost-savings for enterprises by eliminating:

  • The need for local PSTN gateways from costly separate voice ISDN BRIs (Basic Rate Interfaces) or PRIs (Primary Rate Interfaces) and data circuits
  • Multiple voice and data hardware systems
  • Separate network management tools
  • Conferencing and webseminar bridge services
  • Domestic and international long distance charges
  • Security risks through voice encryption
  • Duplicate trunks for disaster backup (or rather add additional redundancy via multiple SIP gateways)
  • Need to terminate calls via PSTN by using E.164 ENUM (internet telephone numbers) services
  • And other organizational management costs.



While SIP brings advancement in VoIP call connections, SIP faces the same security attacks as other IP protocols such as HTTP and SMTP such as malformed message attacks, SPIT-SPam over Internet Telephony, buffer overflow attacks, DOS-Denial-of Service attacks, eavesdropping, hijacking, injection of malicious RTP packets into existing RTP flows and other known and yet to be created attacks.   Special SIP firewall and other protection systems are recommended.

Advanced techniques such as SRTP (Secure Real-Time Protocol) is one method to provide an additional level of security. SRTP is a Transport Layer 4 protocol which intercepts RTP packets and then forwards an equivalent SRTP packet on the sending port, and intercepts SRTP packets and passes an equivalent RTP packet up the stack on the receiving port. A single MKI-Master Key Identifier provides digital keys for confidentiality and integrity protection for both for the SRTP stream and the corresponding SRTCP stream.  In addition, salting bits (keys) (added like adding salt to food) can be added to the MKI-Master Key Identifier to protect against pre-computation and time-memory trade of cipher/hacker attacks. This creates “hash code” (like chopped corn-beef hash) unreadable code characters with a nonce (time stamp or other randomly generated code or word). The MKI “salting” guarantees security against off-line key-collision attacks on the key derivation that might otherwise reduce the effective key size. IETF RFC-3711 is recommended for further reading.


SIP Trunking & Old PBX-Private Branch Exchange Systems

One of the advantages of hosted VoIP service is that they can be added without “forklifting out” the existing PBX telephone system. However, the key point is that SIP Trunking and Hosted VoIP or hosted PBX are compatible strategies for user implementation. That is, SIP Trunking can be used at larger locations with hosted VOIP used for virtual or mobile users for maximum benefit and lower TCO-Total Cost of Ownership.

One of the inherent problems to expand a PBX is the need to add station line cards, trunk cards and other equipment are often required. That is, additional PBX equipment is added in a capital-intensive stair-step fashion. This often leads to under-utilized hardware. SIP Trunking eliminates the need for onsite installation and can expand inexpensively, rapidly and remotely.


Road Map

Development of new SIP trunking software applications may provide even more significant savings.

Lastly, SIP Trunking also opens up a vast array of new call processing concepts not yet developed.


SIP Applications & Future Outlook

IM-Call Screening “presence” features are enhanced by SIP Trunking.

“Event” notification can also be enhanced with SIP Trunking for fire-public safety or business applications such as sporting-concert events, restaurant-airline seat availability or stock price monitors. On demand business meetings, training, broadcast announcements, call-to-meeting notifications, even reverse E911 are enhanced with SIP trunking. Integration of additional “third-party” developed SIP-enhanced services provides additional business and enterprise justification for SIP trunking.

SIP Trunking supports next-generation communications service provider applications such as automated (auto-dialing) outbound voice auto-dialed telemarketing, “event broadcast” emails/vmail or inbound touchtone order fulfillment. SIP Trunking supports on-net toll-free calling and conference calling. Inbound or outbound call centers can be connected for normal or overflow call processing. SIP Trunking supports on-net toll-free calling and conference calling. Inbound or outbound call centers can be connected via VTL-Virtual Tie-Lines for normal or overflow call processing.

Coming soon is IMS or IP Multimedia Subsystems the recognized international standard for IP interoperability, roaming, bearer user control, charging and security. SIP is one of the key protocols in an IMS system.

Affordable Colocation and Cloud Sevices
Skip to content