SipTheeSkype Skype SIP Gateway Blog

Latest changes

• Fix codec load failure message


• Added GSM codec - you need to get tritonus_gsm-0.3.6.jar and tritonus_share-0.3.6.jar from tritonus.org and put in same folder

Latest changes

• Change to use common format clip files. All audio file clips now must be 16khz 16bit mono wav files.


• Added codec PCMA (a-law) support - see sample config file


• You can now add your own codecs by implementing local.ua.sscodecs.SSCodec interface


• Ability to configure multiple codecs - see sample config file


• Fix crash if placed on hold then off hold if auth sequence used.


• Cleanup some funky dtmf sdp code


• New parameter noRtpReceivedAutoHangupSeconds to help with silence suppression causing hangups


• obsoleted parameters
  audio_sample_rate
  audio_sample_size
  audio_avp
  


• Fix intermittant problem with Asterisk rtp connection reset by peer message which causes immediate hangup


• Asterisk compatibility enhancements canreinvite=yes now works (for me anyway)

Latest changes:

• Fix some exception handlers


• Minor changes to some messages

Since I blew away all previous comments from moving around. Here is the change log history.

2008/08/23
Get rid of warn errors due to received sip messages before initialize has completed
Added new params to help with nat issues
sendResponseUsingOutboundProxy
useViaRport
useViaReceived
Change to call logging
make sure to overwrite log.properties

2008/08/18
Fix send dtmf to skype bug
New parameter enableSendRTPtoReceivedAddress to help with one way audio

2008/08/14
Fix wrong underrun count when on SIP hold
Some cleanup


2008/08/13
Fix intermittent speedup noise
More RTP enhancements


2008/08/12
Added Freeswitch reg example
Some comment changes/additions
New parameter audioPriorityIncrease
Modified audio speedup code to be variable
Added more audio stats
Fix connector load library error display


2008/08/09
Added load lib test if skype4java fails to load connector so the real error is seen
Change to startup scripts to specify lib path for connector libraries
Connector libraries have been extracted into the SipTheeSkype root
Change behavior of maxAudioDelayMs - it will double speed the overage, if it too far behind resort to chopping
Added display of chopping,compress, underruns stats if any performed.
Added new Parameter useskypetimingsource
Added different status codes depending on skype call failure reason
603 Refused
404 Failed, Invalid user, no skype credit (Can't tell the difference)
408 UNPLACED whatever that means
600 Busy
403 Anything else


2008/08/05
Renamed to SipTheeSkype
manual compile script fixes
removed extraneous stuff
updated sample config comments

2008/08/03
Added new param maxAudioDelayMs
Fixed All Channels full 486 response ACK loop bug

2008/08/02
added manual build scripts - linux version no tested at all
fix registration warn messages
quiet down the console when not in debug mode

Added SIP Inband DTMF detection

new params:
SkypeDtmfDetectorHitThreshold
SkypeDtmfDetectorSilenceThreshold
enableSIPInbandDtmfDetector
SipDtmfDetectorHitThreshold
SipDtmfDetectorSilenceThreshold
SipOutDialingRulesFile

SkypeDTMF Detection is now working. To activate, set:
enableSkypeDtmfDetector=yes

For SIP Inband, set:
enableSIPInbandDtmfDetector=yes
You should disable rfc2833 and info in this mode



2008/07/28
debugging changes


2008/07/25
Fixed call logging
Fixed dialing rules file case
New param logConfigFile
RTP enhancements


2008/07/22 - first version of v3 - limited testing
---- You must add some new parameters if upgrading from v2----
Changed linux startup script to linux/unix format
Fix EarlyMedia with pin issue.
Added exit code so scripts can detect startup failure.
Parameters skype_audioinport and skype_audiooutport merged to new single parameter skype_audioportbase
New parameter concurrentcalllimit
New parameter skypeclientsupportsmulticalls
Refactored to handle multiple call streams - Win Skype Clients 3.6 - 3.8 do not support multi active calls - do any?
A non supporting skype client will show on console "A call exists in INPROGRESS" if attempting multiple outbound calls.
Now supports SIP Hold
Now supports Skype Hold
Changed to skype4java version as of 2008/07/14
New parameter SkypeOutDialingRules - set your area dialing preferences - look at cfg file for example
New parameter sendSipDtmfToSkype
New parameter sendSkypeDtmfToSip - not working
Added 401/407 INVITE Cancel patch
New parameter SkypeInboundAllChannelsBusyAction - what to do when no more available channels

2008/06/28
Added new parameter autoShutdownMinutes - shutdown after time has expired and no call in progress - 5 minute increments.
Added support for DTMF over INFO messages.
Added new parameter dtmfinfotype - set to application/dtmf or application/dtmf-relay to activate DTMF over INFO msgs.
added enableSkypeDtmfDetector to allow experiments with decoding of skype analog dtmf
Fixed blank pin not working

2008/06/16
moved OPTIONS handler code to a earlier location to guarantee message gets intercepted.
Fixed some more java 1.5 compatibility issues.
Added 486 response to invite if no INVITE listener available.
Added new parameter skypeimmessage to set your own IM message.
Added init contact info if none specified.

2008/06/14
Added new parameter dtmf2833payloadtype - default is still 101
Change RTP delay timing to adjust for cpu loading - affects playing of files and authentication only
Changed blank RTP packets to contain a valid value.
Fix some issues with SIP dtmf sending code.
Added cnonce/nc support if qop authentication specified
Reduce java requirements to java 1.5
Fix intermittant loss of INVITE handler - It would stop answering SIP calls
Refuse multiple incoming skype calls since it isn't supported.

2008/06/06
Added SIP OPTIONS message support so Asterisk can determine status
Reduced likelihood of getting a duplicate branch
Fix REGISTER www authenication not creating new branch
Fix INVITE www/proxy authenication not creating new branch
Added new parameter sendSkypeIM to send an incoming call message when calling a skype user

2008/06/01
Removed setting of skype userid in User Agent

2008/05/28
Fixed linux/mac startup script.
New experiental handleEarlyMedia flag to send Skype status audio during SkypeOut calls


2008/04/27:
added ack 401/407 invite response to help with asterisk 491 error

2008/04/06:
Change to mjua to reset local session when handling calls - it was using last incoming SIP call origins as it's own for outgoing SIP calls.
Changed origin to set a session id. removed user name prefix from origin owner
Fixed real registration problem. When provider sends multiple expires tags, mjsip was grabbing the wrong one. Changed to select the highest which should be the most recent registration.


2008/03/18:
Updated call logging format
Registration mystery solved. Provider sending retarded expiration times.
Two new parameters added to help with funky registration issues.
Added minregrenewtime to cfg file - Default is 60 seconds. A warning message will be put in log if provider tries to set below this limit.
Added regfailretrytime to cfg file - Default is 15 seconds. If a registration fails. It will retry again in this number of seconds instead of waiting the normal expires time.

2008/03/16:
Added function to detect when queued clips are done playing
Added ability to play files to skype callers
Changed to latest skype4java. Fixed dropped skype calls not being detected
Got rid of my pipeinputstreamsizable since jre 1.6 has sizable pipes now.
Increased pipe buffer sizes - They were getting overrun under load.

2008/03/13:
Added RFC2833 DTMF sending to SIP caller.
Added pin/destination retry limits.
Added ability to change pin/dest clip files.
Throw away any inbound rtp packets received before we are ready to process.
Added ability to send a dtmf to sip caller on sip accept (i.e. grandcentral, etc.)
Added new pause and dtmf commands to SipToSkypeAuth.props.
Modification for playing of multiple files syntax - you now have to separate each file to play. Example: play:filename1;play:filename2;
Added missing libraries for Skype4Java

2008/03/10:
Change siptoskypeauthmapper to cycle through the list in sequence.
Fixed bug with id pattern.
Removed double buffering of audio.
Changed logging to log4j - needs log4j-1.2.15, logging controlled by log.properties.
Added logging of Sip To Skype calls.
Allow pintimeout & destinationtimeout to be set in config file.
Delay auth start until call is confirmed by mjsip.
Include skype_full.jar instead of just win32.

2008/03/02:
Fixed mjsip sdp parser problem with c parameter not above first media descriptor
Fixed dtmf map not being removed if remote doesn't support it.

2008/03/01:
Added RFC2833 DTMF detection from SIP side.
Added Pin authorization and destination entry via DTMF - incoming sip callers
Added audioClip playback to incoming sip callers
updated explanations in SipToSkypeAuth.props and SkypeToSipAuth.props
Allow // to be indicate comments after parameters of SipToSkypeAuth.props and SkypeToSipAuth.props files

2008/02/23:
Added auto hangup if loss of incoming rtp for 5 seconds

2008/02/21:
Remove redirect_to and replaced with new authorization system
see SipToSkypeAuth.props and SkypeToSipAuth.props
Got rid of annoying sounds playing on computer

2008/02/20:
Fixed registration bug.
Updated dial plan in readme to use # not * to avoid conflict.
Added registration example in cfg file.