s1p

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Your rating: None Average: 4.9 (16 votes)

s1p is an experimental SIP client for Sailfish OS

Successfully tested with:

  • Asterisk
  • sipgate
  • sipnet.ru
  • linphone.org
  • 3CX
  • cellip.com
  • FritzBox

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

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

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

 

Screenshots: 

Keywords:

Application versions: 
AttachmentSizeDate
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
File harbour-s1p-0.3.1-1.armv7hl.rpm2.18 MB04/07/2020 - 12:42
File harbour-s1p-0.3.2-1.armv7hl.rpm2.18 MB08/07/2020 - 23:56
File harbour-s1p-0.3.3-1.armv7hl.rpm2.18 MB09/07/2020 - 22:58
File harbour-s1p-0.3.4-1.armv7hl.rpm2.19 MB12/07/2020 - 12:46
File harbour-s1p-0.3.5-1.armv7hl.rpm2.19 MB15/07/2020 - 02:04
File harbour-s1p-0.3.7-1.armv7hl.rpm2.19 MB16/07/2020 - 10:34
File harbour-s1p-0.3.8-1.armv7hl.rpm2.19 MB18/07/2020 - 11:18
File harbour-s1p-0.3.9-1.armv7hl.rpm2.2 MB23/07/2020 - 17:04
File harbour-s1p-0.4.0-1.armv7hl.rpm2.2 MB25/07/2020 - 10:52
File harbour-s1p-0.4.1-1.armv7hl.rpm2.2 MB26/07/2020 - 12:06
File harbour-s1p-0.4.2-1.armv7hl.rpm2.2 MB27/07/2020 - 15:43
File harbour-s1p-0.4.3-1.armv7hl.rpm2.2 MB30/07/2020 - 22:23
File harbour-s1p-0.4.4-1.armv7hl.rpm2.2 MB31/07/2020 - 14:47
File harbour-s1p-0.4.5-1.armv7hl.rpm2.2 MB01/08/2020 - 11:01
File harbour-s1p-0.4.6-1.armv7hl.rpm2.2 MB01/08/2020 - 23:13
File harbour-s1p-0.4.7-1.armv7hl.rpm2.2 MB04/08/2020 - 10:53
File harbour-s1p-0.4.8-1.armv7hl.rpm2.2 MB06/08/2020 - 09:50
File harbour-s1p-0.4.9-1.armv7hl.rpm2.2 MB06/08/2020 - 14:06
File harbour-s1p-0.5.0-1.armv7hl.rpm2.2 MB10/08/2020 - 15:21
File harbour-s1p-0.5.1-1.armv7hl.rpm2.2 MB10/08/2020 - 15:28
File harbour-s1p-0.5.2-1.armv7hl.rpm2.2 MB13/08/2020 - 11:57
File harbour-s1p-0.5.3-1.armv7hl.rpm2.2 MB13/08/2020 - 12:08
Changelog: 

- 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

Comments

defactofactotum's picture

Still no luck, the settings info is lost every time and no registration on XA2. On a newly reset X it's the same, but thanks for helping....

unmaintained's picture

You could try to delete s1p configuration completely and start from scratch:

rm /home/nemo/.config/harbour-s1p/settings.json

I've been able to succesfully register with sip.linphone.org so this may be more of an issue with configuration than with registration.

 

 

amaretzek's picture

Doesn't work via UI. It works if you edit the file. Verified on a fresh install on JP1 v3.2. Also, latency and buffers should come with defaults.

defactofactotum's picture

I set the register frequency but it makes no difference. When I change settings I press 'Accept' and it returns to the dialler page as 'not registered'. In the past it said registering and even authenticating but never succeeded. Closing the app changes nothing for me. Just in case this is a linphone thing, can someone recommend an easy to use alternative sip?

unmaintained's picture

please check if version 0.3.2 works with linphone.org

defactofactotum's picture

Hi, recent builds do not register for me with linphone (if anyone has done this please post details) and does not save settings after I press accept - it returns to not registered and settings is again blank. Help! It would be great to get this working.

unmaintained's picture

Is "Regsiter frequency" set to somethig between 3600 and 60?

Also don't forget to close the app after changing settings. 

amaretzek's picture

feature request:server reachable indicator. Or the other way round, "registered" turns red if server becomes unreachable....

unmaintained's picture

It should already indicate if registering failed but the default regstration frequency is 1h or so, means it can take while. s1p does not send any options packets yet.

amaretzek's picture

For 0.3.1 "registered" never changes, event if "regstration frequency" (which is an interval, right??) expired after loss of network. Also, the green phone gets active if you type a number, which is misleading, since there is no network. No problem for me, but for a normal user...

unmaintained's picture

After the regsitration expires nrad should re-register but based solely on time not on any changes to the network.

amaretzek's picture

Yeah. But, if register frequency is 70, presume seconds, if I wait 2 min after loosing network, I should see something. And I didn't. Somehow, I didn't wait long enough. After a while it was "registering". This is ok, but still the green phone was active while it didn't register due to lack of network. It got registered as soon as there was network, as expected. To me this means, ready to test with normal user.. Thanks a lot!

unmaintained's picture

What I mean is that if the green button is enabled or not does not depend on the registration status.

amaretzek's picture

But should, my 2 cents. The green button gets activated when a digit is typed. In my opinion, this should work the same for "server reachable" / "not reachable".

unmaintained's picture

Yes, it ets activated because without any destination to call there is no point in trying to send a sip invite but there's no way to know if at that moment the sip server is reachable or not without actively sending a packet.

amaretzek's picture

It always sends traffic through wlan0? I configured a server reachable via tun0, but packets get out on wlan0, disregarding kernel routing table.

unmaintained's picture

It's not designed to do that but maybe it has something to do which IP address it binds to? Could you check with the log if this would make sense?

amaretzek's picture

netstat says so. Why not binding to 0.0.0.0?

unmaintained's picture

I wonder if it would work if you would change the static IP address in /usr/share/harbour-s1p/qml/pages/s1p.py
line 187 (in function get_own_ip) to the IP of your SIP server (or VPN endpoint).

amaretzek's picture

Yes, it works for registering, didn't try calling, will do it later on. But I don't expect surprises... I used the VPN endpoint (on tun1).

amaretzek's picture

sip audio ok...

unmaintained's picture

Please check if this issue has been fixed in 0.2.5

amaretzek's picture

Yes, it works, at least for registering. Presume audio is ok, will check later on (tomorrow). Thanks!

unmaintained's picture

It will try to bind to the same IP for signalling and audio so I would expect ist to ether both to work or both to fail.

mano's picture

Power consumption: Hello, thanks again for taking the challange to complete the phone capabilities of Sailfish! And for this great user support!
Unfortunately I noticed a hughe battery drain on my Xperia X.
AP Infrastructure is constant for over a year here and keeping mails in sync over WiFi always needed about 30% capacity from (early) morning to (late) evening. Nothing changed in that regard, just started s1p in additon. Now I need 80% of the battery's capacity during the same time frame.
Lighthouse shows average but constant 3% of CPU usage for the s1p process.
SysMon indicates that CPUs are kept from idling and most likely even from reducing clock – but I can't verify that.
Is the power consumption known to be a fixable/optimizable problem in s1p's current state?

Thanks,
-mano

unmaintained's picture

I guess it should be fixable once I've been able to locate the cause :) I do have a few suspects but didn't have the time to look into it.

amaretzek's picture

Power consumption on my Xperia X starts after a call. Before it is ok.

unmaintained's picture

That could happen if e.g. the PulseAudio handler would not shut down properly after a call.

amaretzek's picture

Might have changed. ATM top says 12% CPU for s1p and 7% pa after start, 35%/7% during call, then back to 12/7 after call. Will look at battery drain in one hour or so.

unmaintained's picture

That could still be the PulseAudio handler. I've changed the way starting/stopping PA streams works so PA streams would not run amok after being stopped but this may use more cpu overall.
I will definitely look into it at some point. If I can make them stop when they are supposed to be stopped there should not be much left that could keep the cpu awake.

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