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Wiki HomeSoundwin Provide Free Embedded H.323 Gatekeeper for registering and visiting
http://61.218.109.82
username and password e-mail request sam@soundwin.com for security reason
Soundwin is a VoIP Equioment manufacturer
VoIP Feature:
- Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO
- Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name/IP address, PSTN or PBX) or hunting number setting.
- AC termination Impedance : 600/900 OHM and complex impedance
- Polarity Reversal Detection: Type I and Type II
- passNAT: This feature allow gateway to operate behind any NAT/Firewall device. There is no need to change any configuration of NAT/Firewall like setting virtual server.
- Smart-QoS Guaranteed: This bandwidth management feature provide good voice quality when user place VoIP call and access internet at the same time. The gateway will start reserve bandwidth for voice traffic automatically when VoIP call proceeds.
- Voice channels status display: This function display each port status like as onhook, offhook,calling number callee’s number, talk duration, codec.
- H.323 MAC authentication : Providing H.323 MAC authentication to register H.323 Gatekeeper which need Mac address for authentication. (Note : Soundwin’s Embedded H.323 Gatekeeper provides IP address, H.323 ID, MAC address authentication policy)
Page Changes
Decent FXO and FXS to SIP interface
After facing horible echo problems with Sipura SPA-3000 and Digium cards on some networks, echo was a big concern on any FXO interface. The Soundwin has a 64ms echo cancellor (EC) that does a great job at EC. I've been testing it on a couple of POTS lines that no amount of tweaking on the Digium card could clean up. Thus far not even the slightest echo has been heard.
As an FXO port it has a couple of rough edges. Auto answer cannot be shut off, it can only be delayed up to 8 seconds. When it answers the call it plays an "answer tone" and then connects to the registered SIP server. This tone can be shut off, but it then plays a very short dial tone in its place. There is currently no way to make it simply ring the SIP server and wait for an answer before picking up the line.
When using g729 the comfort noise generator seems way too loud. Every time you talk you hear excessive static from the EC. Under ulaw it seems fine.
So far this is the best bang for the buck FXO port I've used.
They have everything from a single FXS & FXO gateway all the way up to 24 port versions. Each size is available 100% FXO, 100% FXS and 50/50 FXO/FXS.
UPDATE: (5/27/06) After trying it on a production system for a few days some significant problems have surfaced.
1. The echo cancellor has a significant problem with double talk. Instead of diverging it seems to over mute one or both sides.
2. After running for a while it will fail to authenticate with Asterisk when answering a call and play a busy signal to the caller. Switching to an unauthenticated stanza in sip.conf is the only thing that seems to keep calls coming in.
3. Periodically calls seem to be spontaniously dropped.
I have pulled it from the production system and all lines are connected to the highly reliable and predictable Sangoma A200d.