Thursday, 16 July 2015

BEOX-,High Quality VoIP Routes,VoIP Interconnection,Sell VoIP Wholesale Routes,VoIP Calls Services,High Quality VoIP Routes,A-Z VoIP Direct,A-Z VoIP Carriers Providers,A-Z VoIP Direct Routes


eox  helps providers and carriers from all segments of the industry succeed with a range of international voice termination products that provide A-Z coverage and benchmarks  to fit your requirements.

CLI Assured Voice provides the highest quality A-Z voice termination via PTT bilateral routes, with higher-than-market-level Average Call Duration (ACD), Answer Seizure Ratio (ASR) and Network Effectiveness Ratio (NER).  Guaranteed CLI and roaming and lower-than-market level Post Dial Delay (PDD) provide an exceptional service with the highest possible quality/price ratio in this segment.

Beox Certified Voice delivers end-to-end MVNO quality direct coverage to an increasing number of key fixed and mobile networks.  Beox certified routes are selected and actively managed by Beox to meet stability, capacity and voice quality demands of retail operators at very competitive rates.  The code coverage is augmented by our CLI Assured routes to provide a truly A-Z product.

Designed primarily for cost-driven retail operators and resellers, Wholesale Voice features our direct  and partner routes and, offers A-Z coverage at our most aggressive pricing. Wholesale Voice combines extensive code coverage and advanced routing management to deliver stable quality and great value.

Subject to the business case, Beox could design and price a termination solution that gives the best fit your specific quality requirements.

Sales Enquiry : sales@beox.com

Network elements[edit]

SIP defines user-agents as well as several types of server network elements. Two SIP endpoints can communicate without any intervening SIP infrastructure. However, this approach is often impractical for a public service, which needs directory services to locate available nodes on the network.

User Agent[edit]

SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction.[7]
A SIP phone is an IP phone that implements SIP user agent and server functions, which provide the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer.[12][13] SIP phones may be implemented as a hardware device or as a softphone. As vendors increasingly implement SIP as a standard telephony platform, often driven by 4G efforts, the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. Examples are devices from Nokia and BlackBerry.[14]
In SIP, as in HTTP, the user agent may identify itself using a message header field 'User-Agent', containing a text description of the software/hardware/product involved. The User-Agent field is sent in request messages, which means that the receiving SIP server can see this information. SIP network elements sometimes store this information,[15] and it can be useful in diagnosing SIP compatibility problems.






























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