"arbitrary" setting of SIP destination via chat message

Description: this option allows to set SIP destinatation for incoming Skype calls.
Destinaiton can be changed by chat message. This feature gives unique flexibility it call routing.

"arbitrary" mapping of SIP names to Skype names

Description: this option allows to "map" SIP destination username to Skype username. For example (having this option enabled) if SIP endpoint calls PSGw (running at IP address 192.168.0.1) like "myskypeintheoffice@192.168.0.1" then PSGw will use "myskypeintheoffice" as Skype destination.

DTMF bypass from Skype to SIP and from SIP to Skype

Description: digits in chat messages from Skype remote party are sent as SIP DTMF. This allows to use IVR, perform DTMF authorization and use other services that require DTMF signals. PSGw allows to send DTMF to SIP as RFC2833, SIP INFO with String or SIP INFO with tone.
All DTMFs are recognized by Asterisk.

better firewall/NAT support in "Advanced Options"

Description: allows to use STUN server or direct IP address mapping to bypass firewall or NAT.

caller's Skype username displaying as SIP phone with custom prefix

Description: Skype username of calling party is displayed at SIP phone or application. It is possible to add custome prefix.
For example: when Skype user "officeman" calls SIP phone then SIP phone shows "Skype:officeman" as caller's username.

codec management

Description: allows to enable or disable codecs used for SIP calls.

Skype:	PSGw uses preinstalled Skype client and Skype API. Thus it is necessary to have Skype running and logged.
SIP:	PSGw is compatible with most of SIP applications and devices. PGw supports SIP proxies and SIP registrars with authorization.

PSGw can handle only 1 (ONE) simultaneous call due to the fact that Skype can handle only one call.

Images with typical PSGw configuration for single computer is available here, for enterprise configuration here.
Also avaliable PDF White Papers.

Other requirements:

• 10 MB hard disk space
• 256 MB memory
• Windows 2000/XP
• Skype 3.X

SIP module features:

• registrar support
• proxy support
• authorization for proxy and registrar
• external IP address mapping support
• "arbitrary" setting of SIP destination via chat message
• "arbitrary" mapping of SIP names to Skype names
• DTMF bypass (as RFC2833 or INFO) from Skype to SIP and vice versa
• displaying Skype caller for SIP remote party
• STUN server support
• adjustable jitter buffer
• Asterisk compatible

Codecs:

• G.711-uLaw-64k
• G.711-ALaw-64k
• Speex
• G.726
• iLBC
• GSM

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. Silence detection
    This options works with SIP. It allows to minimize usage of bandwidth when there is no talking at PSGw side.
    
    
11. Jitter buffer
    Jitter buffer is used to mimizime the influence of network conditions. It also helps to maintain the voice quality.
    
    
12. STUN server
    STUN server helps to bypass firewall or NAT (Network Address Translation).
    
    
13. SIP User-Agent
    This feature allows to change SIP User-Agent field.
    
    
14. Prefix for Skype->SIP shown name
    This feature allows to add prefix to Skype username shown at SIP 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 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 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.
    
    
17. 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.
    
    

PSGw 3.5 uses SIP protocol version 2.0 and is compatible with most of SIP applications and devices.
Note for Asterisk users: PSGw 3.5 is compatible with Asterisk.


"General Options" screen:




"SIP Parameters" screen:




"Routing Table" screen:




"Advanced Options" screen: