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New: PSGw for Linux 1.5 released - more information...

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PSGw 3.5 Pro (SIP and H.323 version) details

is available for $49.95

• new SIP and H.323 stack gives stability and better voice quality
• improved interaction with Skype
• "arbitrary" setting of SIP/H.323 destination via chat message
• prefix for Skype->SIP/H.323 shown name
• codec management
• selection of DTMF mode for SIP and H.323 (RFC2833, SIP INFO String/Tone, Q.931, H.245 String/Tone)
• compatible with Asterisk

2. Requirements

Typical configuration

Other requirements

• 10 MB hard disk space
• 256 MB memory
• Windows 2000/XP

3. Technical specifications

4. Features

1. Codecs
   PSGw supports codecs:
   1. G.711-ALaw-64k
   2. G.711-uLaw-64k
   3. GSM-06.10
   4. G.726-16k
   5. G.726-24k
   6. G.726-32k
   7. G.726-40k
   8. iLBC-15k2
   9. iLBC-13k3
   10. SpeexWide-20.6k
   11. SpeexNarrow-18.2k
   12. SpeexNarrow-15k
   13. SpeexNarrow-11k
   14. SpeexNarrow-8k
   15. SpeexNarrow-5.95k
   16. LPC-10
   17. MS-IMA-ADPCM
   18. MS-GSM
   
   
   
2. Registar support
   PSGw can run as standalone SIP endpoint. It also can register at SIP Registrar.
   
   
3. Proxy support
   PSGw can use SIP Proxy server to accept and place calls. Very often SIP Registrar works also as SIP Proxy.
   
   
4. Registrar and proxy authorization
   PSGw can use basic SIP authorization to make secure access to SIP Registrar and SIP Proxy server.
   
   
5. Bypass DTMF from Skype to SIP
   PSGw can also detect incoming DTMF signals from Skype. This feature depends on Skype and doesn't work in 100%. PSGw also supports "Bypass DTMF from Skype chat to SIP" feature.
   How it works: remote Skype party can send DTMF signals in chat messages. Each chat message should consist single digit. PSGw detects such chat messages and send them as SIP DTMF (RFC2833) to remote SIP party. This feature should be enabled at "SIP Parameters". This feature also can be used with SkypeIn and SkypeOut.
   
   
6. Bypass DTMF from SIP to Skype
   PSGw can detect infoming SIP DTMF (sent as INFO or RFC2833). When PSGw detects that remote SIP endpoint sends DTMF it sends Skype DTMF to remote Skype party. This feature also can be used with SkypeIn and SkypeOut.
   
   
7. Fixed routing entries
   PSGw has "Routing table" that allows to route incoming calls according to caller's username.
   How it works: PSGw extracts caller's username and selects destination according to entries in "Routing table" .
   
   
8. Map SIP destination to Skype users
   Besides "Routing table" PSGw has powerful feature "Map SIP destination to Skype username" (also called "Arbitrary mapping"). This feature allows to "map" SIP destination username to Skype username.
   How it works: assumed that SIP endpoint calls PSGw (running at IP address 192.168.0.1) with SIP address "myskypeintheoffice@192.168.0.1". PSGw uses "myskypeintheoffice" as Skype destination and connects call to Skype user "myskypeintheoffice". Anything entered before "@" in SIP address will be regarded as Skype username.
   
   
9. Presenting Skype caller username as SIP username
   PSGw can show Skype caller username as its own SIP username.
   How it works: Skype user "OfficeCenter" calls PSGw. According to "Routing Table" or "Arbitrary mapping" PSGw calls remote SIP endpoint. Remote SIP endpoint displays that SIP user "OfficeCenter" is calling.
   
   
10. Send DTMF as
    This feature allows to select how DTMF are sent from Skype to SIP endpoint. PSGw allows to select RFC2833, SIP INFO String or SIP INFO Tone types. RFC2833 requires G.711a/u codec. All DTMFs are recognized by Asterisk.
    
    
11. Silence detection
    This options works with SIP. It allows to minimize usage of bandwidth when there is no talking at PSGw side.
    
    
12. Jitter buffer
    Jitter buffer is used to mimizime the influence of network conditions. It also helps to maintain the voice quality.
    
    
13. STUN server
    STUN server helps to bypass firewall or NAT (Network Address Translation).
    
    
14. SIP User-Agent
    This feature allows to change SIP User-Agent field.
    
    
15. Setting of SIP destination via chat message
    This feature allows to select to change or update SIP destination for incoming Skype calls.
    How it works: Skype user sends message to PSGw with new SIP destination. Then Skype user places Skype call to PSGw. PSGw will route incoming Skype call according to destination defined in chat message.
    
    

1. Codecs
   PSGw supports codecs:
   1. G.711-ALaw-64k
   2. G.711-uLaw-64k
   3. GSM-06.10
   4. G.726-16k
   5. G.726-24k
   6. G.726-32k
   7. G.726-40k
   8. iLBC-15k2
   9. iLBC-13k3
   10. SpeexWide-20.6k
   11. SpeexNarrow-18.2k
   12. SpeexNarrow-15k
   13. SpeexNarrow-11k
   14. SpeexNarrow-8k
   15. SpeexNarrow-5.95k
   16. LPC-10
   17. MS-IMA-ADPCM
   18. MS-GSM
   
   
   
2. Gatekeeper
   PSGw can register at H.323 gatekeeper. PSGw tries to register at gatekeeper when IP address or DNS name is entered.
   
   
3. Gatekeeper H.235 security
   PSGw can use H.235 secure authentication when using H.323 gatekeeper.
   
   
4. FastStart
   PSGw supports H.323 FastStart feature. This feature can decrease time required to establish connection.
   
   
5. Tunneling
   PSGw supports H.323 tunneling feature. This feature can decrease time required to establish connection.
   
   
6. Bypass DTMF from Skype to H.323
   PSGw can also detect incoming DTMF signals from Skype. This feature depends on Skype and doesn't work in 100%. PSGw also supports "Bypass DTMF from Skype chat to H.323" feature. Remote Skype party can send DTMF signals in chat messages. Each chat message should consist single digit. PSGw detects such chat messages and send them as "H.245 Signal String" to remote H.323 party. This feature should be enabled at "H.323 Parameters". This feature also can be used with SkypeIn and SkypeOut.
   
   
7. Bypass DTMF from H.323 to Skype
   PSGw can detect infoming H.323 DTMF. When PSGw detects that remote H.323 endpoint sends DTMF it sends Skype DTMF to remote Skype party. This feature also can be used with SkypeIn and SkypeOut.
   
   
8. Fixed routing entries
   PSGw has "Routing table" that allows to route incoming calls according to caller's username.
   How it works: PSGw extracts caller's username and selects destination according to entries in "Routing table" .
   
   
9. Map H.323 destination to Skype users
   Besides "Routing table" PSGw has powerful feature "Map H.323 destination to Skype username" (also called "Arbitrary mapping"). This feature allows to "map" H.323 destination username to Skype username.
   How it works: assumed that H.323 endpoint calls PSGw (running at IP address 192.168.0.1) with H.323 address "myskypeintheoffice@192.168.0.1". PSGw uses "myskypeintheoffice" as Skype destination and connects call to Skype user "myskypeintheoffice". Anything entered before "@" in H.323 address will be regarded as Skype username.
   
   
10. Presenting Skype caller username as H.323 username
    PSGw can show Skype caller username as its own H.323 username.
    How it works: Skype user "OfficeCenter" calls PSGw. According to "Routing Table" or "Arbitrary mapping" PSGw calls remote H.323 endpoint. Remote H.323 endpoint displays that H.323 user "OfficeCenter" is calling.
    
    
11. Silence detection
    This options works with H.323. It allows to minimize usage of bandwidth when there is no talking at PSGw side.
    
    
12. Jitter buffer
    Jitter buffer is used to mimizime the influence of network conditions. It also helps to maintain the voice quality.
    
    
13. STUN server
    STUN server helps to bypass firewall or NAT (Network Address Translation).
    
    
14. Prefix for Skype->SIP/H.323 shown name
    This feature allows to add prefix to Skype username shown at SIP/H.323 phone or application. This is useful when you want to recognize that remote party is in Skype network.
    
    
15. Codec management
    This feature allows to enable or disable codec for SIP/H.323 communication. You can select either wideband high-quality (G.711) or low-band (GSM) codec.
    
    
16. Send DTMF as
    This feature allows to select how DTMF are sent from Skype to H.323 endpoint. PSGw allows to select RFC2833, H.245 String, H.245 Tone or Q.931 messages. RFC2833 requires G.711a/u codec. All DTMFs are recognized by Asterisk.
    
    
17. Setting of H.323 destination via chat message
    This feature allows to select to change or update H.323 destination for incoming Skype calls.
    How it works: Skype user sends message to PSGw with new H.323 destination. Then Skype user places Skype call to PSGw. PSGw will route incoming Skype call according to destination defined in chat message.
    
    

5. Compatibility

6. Screenshots