Shanghai Baudcom Communication Device Co.Ltd

shanghai, United Arab Emirates

bd-voip-48fxsis a functional analog gateway offering seamless connectivity between voip and traditional phone system. Its a highly-integrated gateway which provides 72 fxs ports with standard r21 interface in 1u size. Users can easily connect their analog phone, fax, pbx via standard voice interface.dag2500 supports the standard sip protocol and its compatible with leading ims/ngn platform and sip-based ip telephony system.

it is ideally suited for small and medium businesses, call centers and multi-location environments that need voip services.

overview:

analog gateway/ippbx locates at the interface between pstn and ip to achieve the conversion between fxo/fxs and voip sipmeanwhile complete the media stream transformation between the bearer channel of pstn and ip.

application:

used for ipcc, ip dispatch, unify communication, voip network, operators, and so on.

specification:

signaling:ss7, pri, r2, cas, q.sig, v5.2, and dpnss;

voip protocol: sip and h.248;

codec: g.711 u-law and a-law, g.711 appendix 1, g.723.1 and g.723.1 annex a, g.729 annex a and annex b, g.726-16g.726-32gsmgsm-efrarmarm-wb, ilbc;

fax protocol:t30, t38, pass through;

network protocol: ip, nat, icmp, arp, http, bootp, ftp, tftp, dhcp, pppoe, snmp, diff-serv;

advanced voice processing technology: voice activity detection (vad), comfort noise generation (cng), echo cancellation, dtmf, caller id, and flexible input/output gain control.

Overview:

Analog Gateway/IPPBX locates at the interface between PSTN and IP to achieve the conversion between FXO/FXS and VoIP SIPmeanwhile complete the media stream transformation between the bearer channel of PSTN and IP.

Application:

Signaling:SS7, PRI, R2, CAS, Q.SIG, V5.2, and DPNSS;

VoIP Protocol: SIP and H.248;

Codec: G.711 U-Law and A-Law, G.711 Appendix 1, G.723.1 and G.723.1 Annex A, G.729 Annex A and Annex B, G.726-16G.726-32GSMGSM-EFRARMARM-WB, ILBC;

Fax Protocol:T30, T38, pass through;

Network Protocol: IP, NAT, ICMP, ARP, HTTP, BOOTP, FTP, TFTP, DHCP, PPPOE, SNMP, Diff-Serv;

Advanced Voice Processing Technology: Voice Activity Detection (VAD), Comfort Noise Generation (CNG), Echo Cancellation, DTMF, Caller ID, and Flexible input/output Gain Control.

overview:

analog gateway/ippbx locates at the interface between pstn and ip to achieve the conversion between fxo/fxs and voip sipmeanwhile complete the media stream transformation between the bearer channel of pstn and ip.

application:

signaling:ss7, pri, r2, cas, q.sig, v5.2, and dpnss;

voip protocol: sip and h.248;

codec: g.711 u-law and a-law, g.711 appendix 1, g.723.1 and g.723.1 annex a, g.729 annex a and annex b, g.726-16g.726-32gsmgsm-efrarmarm-wb, ilbc;

fax protocol:t30, t38, pass through;

network protocol: ip, nat, icmp, arp, http, bootp, ftp, tftp, dhcp, pppoe, snmp, diff-serv;

advanced voice processing technology: voice activity detection (vad), comfort noise generation (cng), echo cancellation, dtmf, caller id, and flexible input/output gain control.

Overview

Baudcom VoIP GSM Gateway is for operators, enterprises, SOHO, virtual operators to offer affordable VoIP solutions. It is a full-featured IP-based GSM wireless network of VoIP gateways, which can provide stable network configuration, powerful features, good voice quality. This GSM Gateway uses the industrys popular chip technology to support E1, GSMCDMA, SIP, IMS interface, supports maximum 4 E1T1, 104 GSMCDMA, 128 SIP trunking. Our GSM gateway is not only full compatible with Asterisk and other leading soft switch, but also is a cost-effective gateway for call termination (VoIP to GSM) and origination (GSM to VoIP).

Overview:

Analog Gateway/IPPBX locates at the interface between PSTN and IP to achieve the conversion between FXO/FXS and VoIP SIPmeanwhile complete the media stream transformation between the bearer channel of PSTN and IP.

Application:

Specification:

Signaling:SS7, PRI, R2, CAS, Q.SIG, V5.2, and DPNSS;

VoIP Protocol: SIP and H.248;

Codec: G.711 U-Law and A-Law, G.711 Appendix 1, G.723.1 and G.723.1 Annex A, G.729 Annex A and Annex B, G.726-16G.726-32GSMGSM-EFRARMARM-WB, ILBC;

Fax Protocol:T30, T38, pass through;

Network Protocol: IP, NAT, ICMP, ARP, HTTP, BOOTP, FTP, TFTP, DHCP, PPPOE, SNMP, Diff-Serv;

Advanced Voice Processing Technology: Voice Activity Detection (VAD), Comfort Noise Generation (CNG), Echo Cancellation, DTMF, Caller ID, and Flexible input/output Gain Control.

overview

bd-e1-sip is cost-effective trunk gateway equipment specific design for open source market to work with asterisk/elastix/small ippbx. Its built-inecho canceller function and voice decoding conversion function. Users can flexibly select g.711a/u, g.729, g.723, ilbc a variety voice decoding anditeasily completesip and pstn voice and signal converting. in addition, bd-e1-sip provides friendly gui configuration interface, users can freely configure and manage the equipment.

overview:

analog gateway/ippbx locates at the interface between pstn and ip to achieve the conversion between fxo/fxs and voip sipmeanwhile complete the media stream transformation between the bearer channel of pstn and ip.

application:

specification:

signaling:ss7, pri, r2, cas, q.sig, v5.2, and dpnss;

voip protocol: sip and h.248;

codec: g.711 u-law and a-law, g.711 appendix 1, g.723.1 and g.723.1 annex a, g.729 annex a and annex b, g.726-16g.726-32gsmgsm-efrarmarm-wb, ilbc;

fax protocol:t30, t38, pass through;

network protocol: ip, nat, icmp, arp, http, bootp, ftp, tftp, dhcp, pppoe, snmp, diff-serv;

advanced voice processing technology: voice activity detection (vad), comfort noise generation (cng), echo cancellation, dtmf, caller id, and flexible input/output gain control.