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.

?  Fax

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.

?  IP

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.

?  Conference

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.

?  Signaling

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  physical interface: conform to the G.703, and support the coaxial cable (75 Ω unbalanced), twisted-pair cable (120 Ω balanced);

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 the T.30;

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

(1) RTP

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.

(2) H.323

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.

(3) SIP

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.

4. SS7

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;

l  …

DONJIN Keygoe200 digital interface multimedia processign board is applied in the medium-sized IP call center.

Solution features:

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.