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SIP = Session Initiation Protocol 15年前已開始開發的 VoIP 語音通話為主之協定, 但這10年間由 H.323 協定主導了 VoIP 發展 (可能是錯誤地主導), H.323 是一種多媒體廣泛協定,  其企圖想包含所有通訊, 圖像, 視像協定, 結果造成了大量不實用封包浪費. 這10年間, H.323的 "假大空" 浪費了全世界大量網絡及硬體資源 .  近年由 SIP 重振雄風, 燃點新一代 VoIP 語音通話之希望 !     VoSIP

SIP = 專; 精; 簡; 快; 聰明之實用語音通話協定,   請按此多看最新 VoSIP 資料

 
H.323 Voice over IP (VoIP) gateway providing real time voice and fax transmission over IP networks. H.323 is specially designed for SME or international corporations who spend vast amount of money on branch-to-branch communication.

By deploying H.323, voice and fax calls, no matter they are made in the office or outside of office, all can be routed through Internet and the long distance calls charges can be eliminated.

Equipped with powerful CPU and DSP, H.323 utilizes advanced compression, echo cancellation, and robust voice and fax recovery algorithm to provide toll quality voice over IP networks. Supporting various types of Internet access, such as ADSL, cable modem and leased line, H.323 can provide value added voice application over the company's existing data network. Easily integrated into any PBX system, H.323 provides immediate cost saving without any impact on the company's telephone system. Much more outstanding features, such as NAT router function, embedded gatekeeper, NAT pass through, bandwidth control etc, make H.323 the ideal VoIP solution for enterprises seeking to significantly reduce their telecommunication cost.

Features and benefits 
     
  Dynamic IP Support  
  The VoIP gateways available in the market today require fixed IP address to determine its location in the Internet. That causes inconvenience to many dial-up and broadband VoIP applications users those don't have fixed IP address. The H.323, however, allows existing addressing schemes to remain without the need of fixed IP address. H.323 can directly connect to the DHCP Enabled cable modem, or PPPoE equipped ADSL modem, which uses dynamic IP address to link to Internet.
     
  Flexible Telephony Interface  
  H.323 provides FXS and FXO interface to be flexibly integrated into any existing telephone system. It can be connected directly to an analog phone, fax machine, key phone system or PBX system. Users can use the original telephone sets on their desk to make VoIP calls as well. It is simple and easy. No additional user training is required.
     
  Quality of Service  
  You may say, "Well, voice over Internet is great, but the voice quality is terrible." It certainly does not apply to H.323. The 802.1p and Type of Service (ToS) feature of H.323 VoIP Gateway guarantees end-to-end voice quality in a managed network. In addition, when interfacing to a very unstable Internet network where it is most likely to encounter a 20% data loss rate, users may hear chopped, low-quality voice. The unique robust voice/fax recovery algorithm of H.323 can largely compensate the data loss rate effect of the Internet and give users a stable voice quality.
     
  Optional Embedded Gatekeeper  
  H.323 has an embedded gatekeeper to simplify the dialing procedure of your VoIP network. This H323 standard gatekeeper supports registration constraints, individual grouping and other basic gatekeeper functions.
     
  NAT Router function  
  This feature allows local network users connect to the Internet using only one public IP address. It supports virtual server, DMZ, packet filtering and DHCP server function. However, if your network already has a router there, H.323 can still be configured to pass through any general NAT router.
     
  Bandwidth Control  
  Providing built-in traffic shaping and bandwidth control functions, H.323 can efficiently control and manage the bandwidth of data traffic by various shaping policies, such as IP address range, service types and priority etc. This considerably enhances the quality of services, especially reducing voice latency, and improving network efficiency without the need of extra expensive traffic management hardware.
     
  4 ports Lan Switch  
  Providing optional built-in 4 ports Lan Switch, can connect four Lan Hubs or Lan switch for a SME or corporate Networks. or direct connect 4 PC for Home use.
     
  Scalability  
  You can flexibly stack a couple of inexpensive H.323 together to obtain a higher port density whenever you need to expand your voice service scale. You don't need to make a huge investment at the beginning of your new venture while the future business remains uncertain. Moreover, in conjunction with our innovative IP address sharing mechanism, you only need a single IP for all the stacked gateways.
     
  Advanced Dialing Plan  
  Advanced dialing plan and address mapping make connection to PABX a simple job. It gives versatile and flexible dialing to any destinations. Moreover, H.323 can be configured to use wildcard as its PABX extensions, significantly simplifying dialing plan for Intranet VoIP applications.
     
  Anti-Line-Seizure Mechanism  
  H.323 supports an intelligent mechanism, patent pending, for detecting line disconnection patterns between FXO interface and PABX or PSTN central office switch.
     
  Management Tools  
  H.323 supports full remote management ability with its embedded Telnet server, HTTP server and SNMP agent. H.323 also provides a real-time "voice call monitoring" interface. This helps the system administrator constantly aware of all the call statistics.
     
  Interoperability  
  H.323 provides full compliance with the H.323 IP telephony standard, ensuring its interoperability with leading third-party hardware and software. H.323 has been proved to be interoperable with Microsoft NetMeeting, Cisco, Lucent, RadVision, NetSpeak and other H.323 v2/v3 compliant equipments..
     
Specification 
     
 
Protocol and Standard LEN Indicators
- ITU H.323 v2/v3/v4 compliant - LAN: 10/100M, Collision, Link, Active
- IETF TCP/IP, RTP, SNMP v2, HTTP,   (Tx/Rx)
  Telnet, DHCP, PPPoE, Dynamic DNS - WAN: Link, Tx, Rx, Collision
    - Line (off-hook/ringing)
Voice Processing - Power
- ITU G.711/64kbps, G.723.1A/5.3,6.3kbps,  
  G.729A/B/8kbps Power
- Voice Activity Detection (VAD) - 100~240V AC, 50~60 Hz to 12V DC,
- Comfort Noise Generation (CNG)   1.2A power adapter
     
Tone Generation and Detection Environmental
- TIA-464B DTMF, Dial, Busy, Ring Back, - Humidity: 10~90%, non-condensing
  Call Progress - Operating temp: 0~50 degrees C
    - Storage temp: -10~70 degrees C
FAX Relay      
- T.30 and T.38 real-time FAX compliant Dimension
- Voice/FAX auto-switch - RG-802: 190mm(L) x 110mm(W) x 35mm(H)
    - RG-804: 240mm(L) x 165mm(W) x 36mm(H)
Echo Cancellation - RG-2500: 440mm(L) x 255mm(W) x 44mm(H)
- G.165/G.168 with 8-16ms echo tail    
  Weight
Telephony Interface   - RG-802/RG-5002: 0.8 Kg
- Flexible combination of FXS and FXO - RG-804/RG-5004: 1.2 Kg
- RG-802/RG-5002: 2 RJ-11 (FXS/FSO) - RG-2500: 4.4 Kg
- RG-804/RG-5004: 4 RJ-11 (FXS/FSO)    
- RG-2500: up to 16 RJ-11 (FXS/FSO)  Safety 
  - UL1950, EN60950
Network Interface    
- 2 Ethernet ports, one for LAN, one for EMC
  WAN - FCC part 15 Class B, CE mark
- 10Base-T and 100Base-T,  
  IEEE802.3 compatible Vibration and Drop
    - IEC 68-2-36, IEC 68-2-6, EC 68-2-32
 

Ports assignment layout sample : any FXS, FXO combination available, 
only this VoIP gateway can fit all your requirement

 

 

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