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s1p is an experimental SIP client for Sailfish OS

Successfully tested with:

  • Asterisk
  • 3CX
  • Yate
  • FritzBox
  • sipgate.de
  • sipgate.co.uk
  • sipnet.ru
  • easyfone.de
  • linphone.org
  • cellip.com
  • easybell.de
  • eventphone.de
  • nfon.com
  • voip.ms
  • voipraider.com
  • peoplefone.de
  • messagenet.com
  • linkspace.it

(please add a comment if you have been able to make it work with a provider not listed above)

Please don't forget to restart the app after making any changes to the settings.

New Salfish versions block access to the contacts database for 3rd party apps. To be able to see your contacts in s1p you have to copy the database file (as root) into the s1p data directory:
devel-su cp -r ~/.local/share/system/privileged/Contacts/qtcontacts-sqlite/contacts.db ~/.local/share/harbour-s1p/




Application versions: 
File harbour-s1p-0.7.9-1.i486.rpm1.85 MB08/10/2020 - 17:24
File harbour-s1p-0.7.9-1.armv7hl.rpm1.82 MB08/10/2020 - 17:24
File harbour-s1p-0.8.0-1.i486.rpm1.85 MB08/10/2020 - 22:46
File harbour-s1p-0.8.0-1.armv7hl.rpm1.82 MB08/10/2020 - 22:46
File harbour-s1p-0.8.1-1.i486.rpm1.85 MB13/10/2020 - 11:40
File harbour-s1p-0.8.1-1.armv7hl.rpm1.82 MB13/10/2020 - 11:40
File harbour-s1p-0.8.2-1.i486.rpm1.86 MB14/10/2020 - 00:37
File harbour-s1p-0.8.2-1.armv7hl.rpm1.82 MB14/10/2020 - 00:37
File harbour-s1p-0.8.3-1.i486.rpm1.86 MB17/10/2020 - 11:16
File harbour-s1p-0.8.3-1.armv7hl.rpm1.82 MB17/10/2020 - 11:16
File harbour-s1p-0.8.4-1.i486.rpm1.9 MB21/10/2020 - 23:00
File harbour-s1p-0.8.4-1.armv7hl.rpm1.87 MB21/10/2020 - 23:00
File harbour-s1p-0.8.5-1.i486.rpm1.9 MB22/10/2020 - 09:43
File harbour-s1p-0.8.5-1.armv7hl.rpm1.87 MB22/10/2020 - 09:43
File harbour-s1p-0.8.6-1.i486.rpm1.9 MB25/10/2020 - 17:08
File harbour-s1p-0.8.6-1.armv7hl.rpm1.87 MB25/10/2020 - 17:08
File harbour-s1p-0.8.7-1.i486.rpm1.9 MB27/10/2020 - 23:32
File harbour-s1p-0.8.7-1.armv7hl.rpm1.87 MB27/10/2020 - 23:32
File harbour-s1p-0.8.8-1.i486.rpm1.91 MB31/10/2020 - 13:41
File harbour-s1p-0.8.8-1.armv7hl.rpm1.87 MB31/10/2020 - 13:41
File harbour-s1p-0.9.0-1.i486.rpm1.91 MB14/11/2020 - 23:12
File harbour-s1p-0.9.0-1.armv7hl.rpm1.88 MB14/11/2020 - 23:12
File harbour-s1p-0.9.1-1.i486.rpm1.92 MB24/11/2020 - 18:57
File harbour-s1p-0.9.1-1.armv7hl.rpm1.88 MB24/11/2020 - 18:57
File harbour-s1p-0.9.2-1.i486.rpm1.92 MB27/11/2020 - 13:14
File harbour-s1p-0.9.2-1.armv7hl.rpm1.88 MB27/11/2020 - 13:14
File harbour-s1p-0.9.3-1.armv7hl.rpm1.88 MB19/03/2021 - 16:46
File harbour-s1p-0.9.3-1.i486.rpm1.93 MB19/03/2021 - 16:46
File harbour-s1p-0.9.4-1.i486.rpm1.93 MB24/03/2021 - 17:31
File harbour-s1p-0.9.4-1.armv7hl.rpm1.88 MB24/03/2021 - 17:31
File harbour-s1p-0.9.5-1.i486.rpm1.93 MB26/03/2021 - 14:35
File harbour-s1p-0.9.5-1.armv7hl.rpm1.89 MB26/03/2021 - 14:35
File harbour-s1p-0.9.5-1.aarch64.rpm1.86 MB13/05/2021 - 13:41
File harbour-s1p-0.9.6-1.i486.rpm1.93 MB13/05/2021 - 18:33
File harbour-s1p-0.9.6-1.armv7hl.rpm1.88 MB13/05/2021 - 18:33
File harbour-s1p-0.9.6-1.aarch64.rpm1.86 MB13/05/2021 - 18:33
File harbour-s1p-0.9.7-1.aarch64.rpm1.86 MB14/07/2021 - 12:02
File harbour-s1p-0.9.7-1.i486.rpm1.93 MB14/07/2021 - 12:02
File harbour-s1p-0.9.7-1.armv7hl.rpm1.88 MB14/07/2021 - 12:02
File harbour-s1p-0.9.8-1.armv7hl.rpm1.88 MB18/07/2021 - 14:32
File harbour-s1p-0.9.8-1.i486.rpm1.93 MB18/07/2021 - 14:32
File harbour-s1p-0.9.8-1.aarch64.rpm1.86 MB18/07/2021 - 14:32

- 0.9.8 allows dialing numbers saved in notes field (and X-SIP field) from contacts.
- 0.9.7 fixes account description, disables issues tracker
- 0.9.6 fixes audio output selection issues introduced with Sailfish
- 0.9.5 added direct access to contacts.db, user accessible database may be not up to date, though
- 0.9.4 minor imporvements to input fields, copying numbers from call history
- 0.9.3 fixes issues with early media
- 0.9.2 fixes some issues with FritzBox
- 0.9.1 fixes some issues with linphone.org
- 0.9.0 fixes default primary account setting
- 0.8.9 adds support for multiple active accounts
- 0.8.8 adds auto-answer
- 0.8.7 opens active account section (instead of always the first section) in settings dialog.
- 0.8.6 adds avatar image for ongoing calls
- 0.8.5 fixes disappearing hangup cover action after a call has been answered
- 0.8.4 fixes double entries in call history, fixes lookup of contacts in call history
- 0.8.3 improves audio routing, removes annoying switch to pre-call audio state with last samples still being played.
- 0.8.2 adds log upload to issues tracker
- 0.8.1 introduces (optional) log file (~/.local/share/harbour-s1p/s1p.log)
- 0.8.0 fixes issues with additional incoming calls during an already establiched call
- 0.7.9 introduces changes to the binary size for faster startup
- 0.7.8 fixes deleting call history, adds remorse timer to delete
- 0.7.7 fixes adding SIM calls to history when disabled, adds experimental issues tracker.
- 0.7.6 adds cellular call history integration
- 0.7.5 adds better notification handling, bringing UI to foreground on incoming calls
- 0.7.4 fixes double notifications
- 0.7.3 adds notifications, changes the way SIP daemon and UI communicate enabling background operations in the future.
- 0.7.2 fixes microphone input with headsets
- 0.7.1 fixes issue with receiving rtp traffic when local IP changes
- 0.7.0 changes to external IP address handling, crash handling, length of call-id and tags
- 0.6.9 fixes screen unlock on incoming calls
- 0.6.8 adds support for hardware/headset buttons to configuration page
- 0.6.7 fixes server port data type
- 0.6.6 adds support for additional SIP accounts (only one at a time can be active)
- 0.6.5 testing FritzBox 7590 compatibility
- 0.6.4 adds less used DTMF digits (A-D,F) to pulley menu, reduces ambiguity with status messages
- 0.6.3 minor visual changes to the cover page
- 0.6.2 adds contact name lookup by phone number
- 0.6.1 adds minor visual improvements to the call history and contacts pages
- 0.6.0 adds minor visual improvements to the contacts page
- 0.5.9 adds voicemail icon and counter
- 0.5.8 adds default audio port selector to settings dialog
- 0.5.7 adds Yate compatibility
- 0.5.6 adds audible ringback tone
- 0.5.5 adds compatibility with Easybell
- 0.5.4 testing compatibility with Easybell
- 0.5.3 fixes issues with saving display names in call history
- 0.5.2 adds display name to call history
- 0.5.1 fixes phone number in history page
- 0.5.0 adds call history page
- 0.4.9 adds small visual improvements to the UI
- 0.4.8 improves compatibility with pjsip
- 0.4.7 fixes error in media description parser
- 0.4.6 improves compatibility with 3CX
- 0.4.5 adds rport option
- 0.4.4 allows to set bind port on the settings page
- 0.4.3 adds support for buttons on wired headsets to allow answering / hanging-up calls
- 0.4.2 adds display activation on incomming calls
- 0.4.1 fixes misleadling log entries related to RTP destination address
- 0.4.0 adds codec selector in settings dialog
- 0.3.9 adds locking down to one of the available codecs when answering a call
- 0.3.8 adds workaround for 3cx
- 0.3.7 reinstates stricter approach to call progess messages
- 0.3.6 allows a more flexible approach to call progess messages
- 0.3.5 fixes proxy-authentication
- 0.3.4 adds support for display name
- 0.3.3 fixes previously broken default settings
- 0.3.2 fixes regsitering with sip.linphone.org
- 0.3.1 sets default register frequency to 1 hour
- 0.3.0 adds options for bind address and regsiter frequency
- 0.2.9 enables voicemail button, fixes issues with number input
- 0.2.8 fixes issues with setting latency and buffer length
- 0.2.7 adds configuration dialog options for latency and audio buffer length
- 0.2.6 adds configuration-file options for latency and audio buffer length
- 0.2.5 adds improvements to power consumption, audio handler and playback buffer
- 0.2.4 adds preferrence for domain instead of IP in SIP dialogs and adds auth-name field to SIP account settings
- 0.2.3 adds initial DTMF support
- 0.2.2 fixes call status being sometimes overwritten by regstration status
- 0.2.1 adds DNS SRV record lookup
- 0.2.0 adds G.711 μ-law codec, (hopefully) fixes some call-state issues
- 0.1.9 fixes choppy audio on some phones
- 0.1.8 makes some changes to how audio frames are handled
- 0.1.7 adds volume presets
- 0.1.6 enables mute button
- 0.1.5 fixes issues with audio output selection
- 0.1.4 enables audio output selection buttons
- 0.1.3 improves logging and handling of audio packets
- 0.1.2 adds log upload
- 0.1.1 adds contacts page
- 0.1.0 improves UI
- 0.0.9 fixes issues with outbound calls through sipgate
- 0.0.8 fixes inbound calls with sipgate (cancelling outbound calls still broken)
- 0.0.7 fixes some issues with sipgate (inbound calls still broken)
- 0.0.6 improves UI and makes hanging up calls more reliable
- 0.0.5 adds ringtones
- 0.0.4 fixes registering issues with antisip.com
- 0.0.3 adds log page
- 0.0.2 initial release


maier's picture

Sorry for the delay. I sent the log now 3x but I can't find an ID at the end. ... it going on and on with logging. Situation don't change. After 1-5 seconds the red button looses focus get dark the green button gets light the red button is not responsible anymore.

unmaintained's picture

Could you please test it again? I've changed the code responsible setting various flags depeinding on the status of the call.

unmaintained's picture

I suspect something must be going on in the background that appears to the clent like the call would have been ended (thus disabling the button) e.g. a second dialog failing while the first is still active. 

maier's picture

Correct. ... I have the feeling that I have to call again to get the red button for a very short time activ.

unmaintained's picture

Could you please test it again with 0.2.0 and, should this issue still occur, send a log and reference the log-id here?

pasko's picture

Thanks again. Keep up the good work!

maier's picture

I installed the latest version. The sound is much better. The UI problems still exist. I have to push the call button before the red button accepts any push behavior. I already sent the log to you.

unmaintained's picture

Does it mean the call gets properly established (you can see the call timer) but the red button remains disabled?

maier's picture

I was playing around and it feels good expect the ending of a call. The UI has no response on the red button. Thanks for your work!

unmaintained's picture

Do you see any errors on the log page that would maybe indicate why this happened?

Bsingleto's picture

Ready to try this out. Couple days ago did a request for SIP access from my voip provider. Today, i enter in name. I enter in pswd, then i try to cut and paste the Server Address line and its un able to paste anything in the field. Whats going on with that? So i try to type each letter/dot in manually and everything on that line erases when i hit (enter) (period) or (forward slash) so I'm unable to see if this app works for me on sailfish os.

unmaintained's picture

As stated in the description there is an issue with resolving domain names. Please use the IP address of the SIP server instead.

defactofactotum's picture

Hi unmaintained, excuse my newbie ignorance, how can I find the IP address of my server (linphone). Could you add this info and an example in the app description? Thanks for your work

unmaintained's picture

I'm sorry, but I don't have any idea how a linphone server operates.

With other services you can often do a DNS lookup of the address:

IP address or host name: sipgate.com

sipgate.com has address

But it SIP can also have a more complicated DNS setup, so this basic approach may not always work.

carlosgonz's picture

Thanks for S1P, It Is free software?

amaretzek's picture

0.1.3: asterisk server on same net, call in and out + audio works! Nice!

naytsyrhc's picture

Thx for this great app and the steady improvements. However, I still have problems with the basic feature of hearing my collocutor. I'm on sipgate.

unmaintained's picture

One-way audio issues are a recurring nightmare on SIP. I suspect it could be NAT related (s1p does not use STUN at the moment)

I've tested sipgate using a public IP address and audio works both ways. When I switch to a private IP address the inbound audio stream seems not to reach the app. But I would strongly suggest you try it out yourself. The way sipgate deals with NAT may play a major role here.

ninepine's picture

Hi. Thanks for the App. Some success on sipgate.co.uk using 9 character SipID (Sipgate seem to have extended my original 7 digit ID with a suffix of e0), Sipgate Password, Server address of and Server Port 5060. The App shows as registered and I was able to initiate a call and send outbound sound. No inbound sound yet. Inbound call to Sipgate number the app shows that a call is being received but the phone (Xa2 running SFOS 3.3) does not ring. Great progress none the less. Thanks for your hard work. Much appreciated.

While I understand that the UI has yet to be developed the Green and Red Call start and end icons are prtially off screen so moving them away from the bottom edge would be good at some point please.

unmaintained's picture

I'm afraid this may take a while. I've tried to register an account with sipgate.co.uk but I have to wait for some kind of security code they are supposed to send me by regular (snail) mail.

unmaintained's picture

I've fixed the issue with keypad buttons touching the lower edge of the screen in version 0.1.0, uploaded literally 5 minutes ago :)

maier's picture

Here it's not working with JP1 and FritzBox. Sorry. The network_ready is not going on.

unmaintained's picture

You could try an earlier version, could be that the last changes broke something.Unfortuantely I don't have a FB to test s1p against. Have any specific error messages been logged?

explit's picture

Fritzbox seems to work basicaly. Don't tested much, but this is a big success for many users in Germany

unmaintained's picture

FritzBox, being an appliance, would also be super hard for me to debug, so I'm really glad to read that.

pasko's picture


Thank you very much for this app. Looks very promising.

I'm willing to help debugging or providing logs if necessary. My configuration includes an Asterisk server with chan-sip with a few clients, including a Linphone client in an Ubuntu box and an Nokia N9 sip client.

I miss the option to use alphabetic characters in the address field from previous versions, since the accounts in my Asterisk server are defined with non-numeric user names.

I have also noticed some timeout problems with stablished calls as logged by the Asterisk server.

Best regards.

unmaintained's picture

As a work around you can use copy & paste to dial non-numeric destinations. I know that's not ideal and I think about how to make it optional. 


pasko's picture


I just noticed the 'paste' icon in the UI. :)

Thank you very much again for your work.

Best regards.

Fly86's picture

There are actually some people including me ;)
Anyway it would be nice if it worked like 1998 :-)

unmaintained's picture

I've added "1998 mode" in 0.1.4  :)