DONJIN Keygoe200 digital interface processing boards adopt PCIe computer bus structure, and one board could provide 2E1 digital trunk interfaces, and two kinds of physical interfaces BNC and RJ48. Using DONJIN Keygoe SDK, It can be used for voice playing/recording, conferencing, faxing, VoIP, ISDN PRI, SS7, FSK data receiving/sending, text to speech (TTS), and many other functions. They can also be used with Keygoe200 analog interface processing board K161A-E. Besides, using the echo suppression processing circuits, the boards have quite excellent voice quality.
l Use advanced Keygoe multimedia processing technology, and inherit the openness and flexibility of the Keygoe series products;
l Be compatible with Keygoe EasyAPI; without any modificaiton, the applciation could be used in all Keygoe series products;
l One board could perform the multimedia processing functions such as the voice, conference, fax, VoIP, and SS7, not requiring to configure any resource board;
l Use echo suppression processing at the interface, and provide the posibility to improve the IP application performance;
l The small size is suitable for space-limited server, PC, or IPC.
? Voice interface
l Adopt special echo suppression processing circuit, and provide a maximum of 128 ms echo suppression processing capability;
l Support the continuous speech processing (CSP), voice interrupt, and the real-time voice playing/recording of memory and file server;
l Support the mix voice playing/recording and voice recording/playing in a full duplex mode;
l Support the conversion between the A law and μ law;
l Possess the function of caller identity detection (CID); support two CID modes: DTMF and FSK;
l Support the DTMF, FSK, and customized envelope tone generation and detection;
l Support to adjust the DTMF receiving/sending sensitivity; adapt to different lines or terminals;
l Support the automatic gain control (AGC/ALS);
l Support the voice activity detection (VAD) and comfort noise generation (CNG);
l Support the DPD and PVD.
l Support to receive/send the .tiff files;
l The supported maximum fax receiving/sending speed rate is up to 14400 bps, and the automatic down-speeding is supported to adapt to various communication environments;
l Support the fax receiving/sending in an ECM mode; during the handshake, the ECM/NON-ECM mode is selected by self-adaptation;
l Support to receive/send a single or multiple fax pages, and the sent/received page count is not limited;
l Support to send multiple files once;
l Support to add customized page headers by function invoking.
l Support the H.323 protocol group, and could connect to the IP telephone gateway, gatekeeper, and soft switch which also support the H.323;
l Support the SIP, and could connect to the function servers that also support the SIP;
l Support encoding/decoding standards, including G.711, G.723.1, G.729A, and so on;
l Support the voice active detection (VAD) and silence compression.
l Detect the DTMF tones generated by any conference member, and shield the tones to other members;
l The automatic gain control (AGC) ensures each conference member has the same volume;
l Personal volume control ensures each member can adjust the volume to a satisfied level.
l Support China SS1;
l Support the DSS1 (ISDN PRI);
l Support the ITU-T SS7 and China SS7; support the MTP, TUP, ISUP, and SCCP (transparency); suppor the point codes of 14 bits and 24 bits.
l Interface number: 2E1;
l Maximum board nubmer in one computer: 2;
l Size: 142 mm * 100 mm;
l Maximum voice resources in one computer: 128 channels;
l Maximum fax resources in one computer: 16 channels;
l Maximum VoIP resources in one computer: 64 channels;
l Maximum conference resources in one computer: 64;
l Maximum SS7 resources in one computer: 8 channels (64 kbps);
l Computer bus: PCIe;
l Operating system: Microsoft Windows XP/2003/2008;
Red Hat Enterprise Linux 5/6, CentOS 5, Debian 6, Ubuntu 11 Server.
1. Voice resources:
l Frame structure: conform to the frame structure of G.704 and multiframe alignment of G.706;
l The 2048 kbps PCM confirms to the G.732 and G.796;
l The alarm conforms to the G.775;
l The jitter and wander conforms to the G.823;
l Voice encoding format: G.711 (support A law/μ law PCM, AMI-ADPCM), G729, G723.1, and so on;
l Voice file format: WAVE, PCM, ADPCM, VOX, and so on.
2. Fax resource
l Support to receive/send the fax confirming to the V17/V29/V27 standards at a rate of 14400/9600/7200/4800/2400 bps;
l Support to receive/send the fax in an ECM mode; during the handshake, the ECM/NON-ECM mode can be selected by self-adaptation;
l Support to receive/send the .tiff file in the MH, MR, and MMR formats.
3. VoIP resource
l RTP/RTCP (RFC3551/3552);
l Voice packet sampling length/frame number can be configured (10–60 ms);
l RTP dynamic DTMF load (RFC2833/4733);
l Jitter buffer: Support the static jitter mode and dynamic self-adaptation mode simultaneously;
l Analyze the network environment and make the statistics of packet loss;
l NAT/firewall detection and traversal.
l ITU-T H.323v2 (H.225v2/H.245v3);
l Call parameters such as the Fast Start, and H.245 Tunneling;
l Call forwarding;
l Gatekeeper auto discovery and login (Support the number registration types of H.323 UID and E164);
l H.245 user input character and DTMF signal message (User Input Indication);
l Terminal registration management.
l IETF SIPv2 standard (RFC3261);
l UDP/TCP call mode;
l MD5(digest) authentication;
l REFER call forwarding (RFC3515);
l SIP (RFC3265 SUBSCRIBE/NOTIFY);
l Proxy server registration, security authentification, and regular refreshing;
l SIP INFO message(RFC2976);
l Registrar; support the authentication of the service part;
l NAT/firewall detection and traversal.
l Conform to the ITU-T signaling system No.7, including the Q.700-Q.716, Q.721-Q.766, and Q.771-Q.795;
l Maximum message processing ability for each link: 500 MSU/S;
l Support the MTP, TUP, ISUP, and SCCP (transparency);
l Support the 64 kbps standard link;
l Support the 14/24-bit point codes;
l Support the traffic balancing between the links and link groups;
l Support the link switch and switch back in a link group;
l Support multiple OPCs (originating point codes) and DPCs(destination point codes);
l Support to dynamically add, delete, activate, recover, deactivate, normally restart, urgently restart signaling and to process signaling congestion;
l Support the SP and STP function.
l Call center of small to medium sized enterprise (including IP call center);
l Telephone exchange of small to medium sized enterprise (including IP PBX);
l Industry IVR;
l Telephone conference system in enterprise level;
l Network fax system in enterprise level;
l Enterprise telephone recording system;
l Dispatching and emergency commanding system;
DONJIN Keygoe200 digital interface multimedia processign board is applied in the medium-sized IP call center.
l Use two K320-E boards or one K640-E board, and use with two K161A-E boards together; provide a maximum of 2 E1 digital interface trunk access and 32 analog agents;
l Support the digital trunk access using the SS7;
l The functions of recording, faxing, IP agent, and conferencing can be performed without configuring any media resource board;
l Adopt special echo surpression circuits for exellent call quality;
l One computer could provide a maximum of 64 IP agents;
l One computer could provide a maximum of 16 fax resources;
l The cost-efficient feature meets the requirment of growth-type enterprise.
Fax resource software
It provides 4 fax resources; support the T.30.
VoIP resource software
It provides 4 VoIP resources; support the SIP and H.323.
SS7 resource software
It provides one SS7 resource; support the MTP, TUP, ISUP, and SCCP (transparency).