s1p

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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. 

 

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
Changelog: 

- 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

c_mauderer's picture

I'll try to do some more analysis. Maybe I can find some reason.

c_mauderer's picture

I did some further analysis: A packet capture of a call using linphone and one with s1p. The most notable difference in the BYE packet is the "To" field. s1p is sending the following

sip:**610@192.168.47.79:5060;transport=UDP

linphone is sending the following:

sip:**610@fritz.box

Note the difference: s1p is sending a port and a ";transport=UDP" together with the address. Beneath that linphone does some address resolution. But I don't think that the address resolution is the relevant part.

You can find screenshots of the complete BYE packets here (for the next month):

s1p: https://nc.c-mauderer.de/index.php/s/GRCm9akECYYKBPJ

linphone: https://nc.c-mauderer.de/index.php/s/EAeqJ9SCcM68f5r

Edit: It seems that I caught the wrong BYE for s1p. That one is in the wrong direction. Please ignore it. I'll search for the correct one.

Edit2: Nope: It was the correct BYE sent from s1p to the fritzbox.

unmaintained's picture

That does not have to mean much at least not if the previous dialogs also used the same uri.

c_mauderer's picture

Thanks for the response. The URI isn't the same for linphone as well as for s1p. See SIP-stream for s1p and SIP stream for linphone.

In both cases the INVITE and ACK used a written name while the BUE used a somehow encoded name.

A big difference is that s1p seems to continue re-registering during the call.

Is there something that I could do to analyze that a bit more? I think your application is currently not open sourced? Would the recorded network streams help? I would prefere to send these via mail instead of a public comment section. So if they would be usefull please drop me a line to contact at c-mauderer.de.

 

unmaintained's picture

Re-Registering happens also during calls like with pretty much any other SIP user agent (but maybe the registartion happens more often), but this is always done as a separate dialog with a completely different Call-ID and should not interfere with active calls.

c_mauderer's picture

Not entirely sure what changed but the problem is solved with 0.2.4. Thank you very much for that.

unmaintained's picture

I'm happy to hear that. I've made some minor adjustmenst to when, which contact information is used and I suspect this to have made all the difference here.

maier's picture

Hi your problem with the calls should be solved now. I use the FritzBox 7490. The problem that you have the speaker on by default I have too but that I have also with the integrated solution from Jolla.

c_mauderer's picture

Hello @maier: No my problems aren't solved. I should have written the versions I'm using: I tried it with s1p 0.2.2 and a FritzOS 07.12 (latest one for the box).

emchella's picture

Great Thank you ..

Enrico

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:
e.g.
https://ping.eu/nslookup/

IP address or host name: sipgate.com

sipgate.com has address 134.119.225.122

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 217.10.79.23 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.

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