VoIP H.323 SDK is based on IETF standards (H.323, RTP/RTCP, STUN, ICE, etc.), so it should be compatible with other standard based products such as: Asterisk. Key Features * Easily make and receive H.323 based phone calls through any H323 gateway or H.323 compliant IP-Telephony service provider * VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law, Speex, GSM6.10, iLBC, LPC-10) * Registration

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H323 phone right at the electronics level. The design is based on Hybrid Combination of RFC SIP Protocol and Windows RTC and is today successfully ported to Windows CE/Pocket PC/Smart Phone. With a memory overhead of 3 MB - the telephony engine very well fits the needs of engineers looking for porting highly advanced SIP or H323 PROTOCOL to the embedded world. Research Lab further enables you to create production level voip system with following

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VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines. Though this implementation might turn out fairly complex, our VOIP Implementation Team at Research-Lab will guide you remotely for the same. VOIP SIP DLL Soft Phone SDK brings SIP protocol support for ActiveX. With this SDK one can create in minutes a VOIP phone program to connect and start speaking with anyone with a direct IP Address

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VoIP (SIP or H323) telephony engine with voice, video, data, instant messaging routing and integrated reporting/billing under one system. TeleFactura also offers access-control with any Radius/AAA-compatible H323, SIP, or PSTN equipment. Supported are: IVR, prepaid, postpaid card system and accounts, call centre, IAX server, and VoIP server, PC2Phone and Phone2PC gateway. These features are offered thanks to a close integration with Yate, Asterisk

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VOIP Phone Buddy for SKYPE VOIP Phone Buddy for Skype adds automatic telephone dialling to almost any windows application, using SKYPE VOIP telephony, including Microsoft CRM, Microsoft, Access, Microsoft Outlook, Goldmine, ACT, other CRM and Accounts packages etc. Simply move the cursor to a phone number and press the Hot Key to activate "VOIP Phone Buddy for Skype" phone dialer. The VOIP Phone Buddy for Skype and Jajah dialer then looks at the

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VoIP Monitor enables you to measure and track performance of voice quality across WAN links. Leveraging Cisco® IP SLAs, VoIP Monitor collects and analyzes VoIP performance statistics including MOS, jitter, network latency, packet loss and other important quality of service (QoS) metrics. These features enable you to proactively find the root cause of VoIP performance degradation and measure expected voice quality in advance of a VoIP deployment.

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