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s1p is an experimental SIP client for Sailfish OS

This is software in an early stage of development - please don't expect it to work with every type of SIP server out there.

It also needs a restart after any changes to the settings.

Successfully tested with Asterisk chan_sip.

Please reference the log-id when writing a comment after submittimg a log. The id should be displayed at the bottom of the page if the log was successfully submitted. 




Application versions: 
File harbour-s1p-0.0.2-1.armv7hl.rpm2.13 MB27/05/2020 - 10:26
File harbour-s1p-0.0.3-1.armv7hl.rpm2.13 MB27/05/2020 - 23:38
File harbour-s1p-0.0.4-1.armv7hl.rpm2.14 MB29/05/2020 - 14:26
File harbour-s1p-0.0.5-1.armv7hl.rpm2.14 MB30/05/2020 - 11:04
File harbour-s1p-0.0.6-1.armv7hl.rpm2.14 MB01/06/2020 - 19:36
File harbour-s1p-0.0.7-1.armv7hl.rpm2.14 MB02/06/2020 - 22:52
File harbour-s1p-0.0.8-1.armv7hl.rpm2.15 MB04/06/2020 - 20:19
File harbour-s1p-0.0.9-1.armv7hl.rpm2.15 MB05/06/2020 - 12:22
File harbour-s1p-0.1.0-1.armv7hl.rpm2.15 MB05/06/2020 - 14:58
File harbour-s1p-0.1.1-2.armv7hl.rpm2.15 MB05/06/2020 - 23:43
File harbour-s1p-0.1.2-1.armv7hl.rpm2.15 MB06/06/2020 - 11:41
File harbour-s1p-0.1.3-1.armv7hl.rpm2.15 MB06/06/2020 - 19:01
File harbour-s1p-0.1.4-1.armv7hl.rpm2.15 MB08/06/2020 - 21:30
File harbour-s1p-0.1.5-1.armv7hl.rpm2.15 MB09/06/2020 - 11:24
File harbour-s1p-0.1.6-1.armv7hl.rpm2.16 MB09/06/2020 - 15:49
File harbour-s1p-0.1.7-1.armv7hl.rpm2.16 MB11/06/2020 - 09:29
File harbour-s1p-0.1.8-1.armv7hl.rpm2.16 MB12/06/2020 - 11:37
File harbour-s1p-0.1.9-1.armv7hl.rpm2.16 MB12/06/2020 - 21:11
File harbour-s1p-0.2.0-1.armv7hl.rpm2.16 MB13/06/2020 - 11:53
File harbour-s1p-0.2.1-1.armv7hl.rpm2.18 MB13/06/2020 - 22:21
File harbour-s1p-0.2.2-1.armv7hl.rpm2.18 MB16/06/2020 - 20:07
File harbour-s1p-0.2.3-1.armv7hl.rpm2.18 MB20/06/2020 - 11:01
File harbour-s1p-0.2.4-1.armv7hl.rpm2.18 MB21/06/2020 - 11:30
File harbour-s1p-0.2.5-1.armv7hl.rpm2.18 MB26/06/2020 - 14:31
File harbour-s1p-0.2.8-1.armv7hl.rpm2.18 MB28/06/2020 - 11:08
File harbour-s1p-0.2.9-1.armv7hl.rpm2.19 MB29/06/2020 - 16:48

- 0.0.2 initial release

- 0.0.3 adds log page

- 0.0.4 fixes registering issues with antisip.com

- 0.0.5 adds ringtones

- 0.0.6 improves UI and makes hanging up calls more reliable

- 0.0.7 fixes some issues with sipgate (inbound calls still broken)

- 0.0.8 fixes inbound calls with sipgate (cancelling outbound calls still broken)

- 0.0.9 fixes issues with outbound calls through sipgate

- 0.1.0 improves UI

- 0.1.1 adds contacts page

- 0.1.2 adds log upload

- 0.1.3 improves logging and handling of audio packets

- 0.1.4 enables audio output selection buttons

- 0.1.5 fixes issues with audio output selection

- 0.1.6 enables mute button

- 0.1.7 adds volume presets

- 0.1.8 makes some changes to how audio frames are handled

- 0.1.9 fixes choppy audio on some phones

- 0.2.0 adds G.711 μ-law codec, (hopefully) fixes some call-state issues

- 0.2.1 adds DNS SRV record lookup

- 0.2.2 fixes call status being sometimes overwritten by regstration status

- 0.2.3 adds initial DTMF support

- 0.2.4 adds preferrence for domain instead of IP in SIP dialogs and adds auth-name field to SIP account settings

- 0.2.5 adds improvements to power consumption, audio handler and playback buffer

- 0.2.6 adds configuration-file options for latency and audio buffer length

- 0.2.7 adds configuration dialog options for latency and audio buffer length

- 0.2.8 fixes issues with setting latency and buffer length

- 0.2.9 enables voicemail button, fixes issues with number input


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  :)

Fly86's picture

Good work.Good working App. Unfortunaly the sound does not come from the earpiece. Would be great if that can still be changed.

unmaintained's picture

You mean people are still holdng their phones to their ears like it's 1998 ?


amaretzek's picture

version 0.0.5, if server is reachable via dev tun, it tries to send via wlan dev. Didn't investigate further.