FAQ

Summary

• Summary
• Before starting
• The other party can hear me but I can not hear them
• Audio not received/sent
• Call stops after 30 seconds
• There is no contact list
• When integrated to android, I don't want some number to be handled by SIP
• Is there a way to close the application?
• On 3G I don't always receive calls
• I receive calls twice / Registration is done on the sip server twice
• I can't end calls
• Audio routing troubleshooting
• When screen goes off sound quality is bad
• In gingerbread with dialer integration I can't place GSM call anymore
• I did a mistake when I set up the voice mail number, how can I modify it?
• I try to use ZRTP + SRTP, but ZRTP doesn't work
• Cryptography notice

For non listed question, do not hesitate to join and ask on the users group :

http://groups.google.com/group/csipsimple-users (csipsimple-users@googlegroups.com).

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Before starting

If you don't know what is SIP and what this program does : WhatIsSIP

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The other party can hear me but I can not hear them

• It's probably a NAT problem. Some networks are NATed network. It means that there is some network equipment between you and the rest of the world that "hide" your IP address to the rest and the world. It results that your SIP client will not announce properly your IP to sip server and other party. So when a call is placed and media stream is established, the remote party doesn't know where to send its stream and you can't hear him.

To solve this problem, you must activate STUN :

Go on setting Settings > Network - Tick "Use Stun" and fill a stun server on the field bellow.

CSipSimple has a default a stun server if you activate STUN option ! (but as soon as you do not activate the STUN setting it is not used), if you want to use another STUN server than the default one, you can see http://www.voip-info.org/wiki/view/STUN Public Stun server section to know what server you could use freely.

You can also try to use ICE in addition to STUN if STUN alone doesn't solve the problem : Settings > Network - Tick "Use ICE"

• Another possible root cause is a problem with Bluetooth feature. Since the app tries to automatically switch to Bluetooth, but that not all devices ROM gives the correct feedback about actual Bluetooth pairing, it may route audio to some virtual device and so you can't talk in the virtual micro :).

In this case, try to disable Bluetooth option of your phone (even if nothing is paired).

• If it doesn't help, there is maybe a problem with your device (on some device, manufacturers does strange things with the android audio stack). So report the problem on the issue section.
• It can also be something with codec negotiation. So it can be interesting to try to disable all codecs but PCMU (aka G711u aka uLaw). To do so, go in settings > media > codecs. And long press each other codecs until you get the popup to tell that you want to disable it.

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Audio not received/sent

If you have problem with no audio on one of the side you can start to eliminate some use case before reporting a problem.

• While in call, you can press the last menu item on the bottom bar that shows a popup with audio activity.

If it's the remote that doesn't ear you, focus on the micro activity :

1. If you see no activity on the micro bar it's probably a problem with the audio layer of the device. You can try to see the instructions here on Audio troubleshooting. If you report an issue, give your device model and the settings your tried in "audio troubleshooting section".
2. If you see activity on the micro bar but the remote doesn't here you, the problem is more likely with network. In this case check the STUN settings on the remote side. It's more likely a problem on remote side in this case. (Note : remote side here can be the actual remote sip client or the sip server in the middle if it's a media gateway).

If it's the app that doesn't produce sound, focus on the speaker activity :

1. If there is activity but you ear nothing, it's probably a problem with the audio layer. Try the Audio troubleshooting instructions and try to change the way the app renders sound (earpiece, speaker, bluetooth, headset). If you report an issue, give your device model, settings you tried in "audio troubleshooting section" and the audio device (earpiece, speaker, bluetooth, headset).
2. If there is no activity here, it's probably because the app doesn't announce a suitable way to reach it in network negotiation. Please have a look to the instructions about about STUN. It might also be something with codec negotiation. You can try to disable all codecs but one you know to be supported by remote side (Note : remote side here can be the actual remote sip client or the sip server in the middle if it's a media gateway).

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Call stops after 30 seconds

This is probably a configuration problem. You should ask your sip provider for settings they expect (particularly whether they expect to be the sip proxy or have a different sip proxy address).

If the call cuts automatically after 30seconds it's because the app never received the confirmation of the fact the call is established. In such situation sip protocol indicates that the call is not valid anymore.

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There is no contact list

Just try to make a call using the standard android phone dialer, or clicking a contact in the official contact application ;).... And you'll see that a contact list is not needed at all in the SIP application !!! There is also a quick text search dialer if you switch to text dialing mode, that will search on android contacts.

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When integrated to android, I don't want some number to be handled by SIP

I want to automatically add prefix/rewrite some numbers

There is a powerful rewriting/filter/auto-answer tool in CSipSimple (see the UsingFilters wiki page). To access this tool, go on the account on which you want to apply things. Press menu button and choose "Filters". You are now in the tool. Using this tool you can :

• Rewrite numbers : for example add prefix, suffix, completely replace the number by another one, or apply a custom regexp - if you don't know what a regexp is, don't use it ;).
• Don't call some numbers : if you don't want some number to be managed by this account, for example call to other mobiles phone etc, you can add an exclude rule. Then when you'll dial from native dialer/contact app if the phone number match this rule, this sip account will not be proposed in the list of choices you have and if there is no SIP account that can handle this number, it will automatically use Mobile without asking you anything
• Force call some numbers using this account : same thing that the last point, but here it will not exclude, but force the call to use this account and will not ask you for anything.
• Auto answer : if you want to automatically answer to some phone number, you can add a rule here. It's useful if you are using some call callback feature from your provider.

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Is there a way to close the application?

Yes, but you have to configure CSipSimple according to the fact you want to be able to close it !

Let me explain : CSipSimple can run under several setting configuration :

• "Always available" when you want to be able to receive calls
• "Only for outgoing calls" when you only want to use sip for outgoing calls (CSipSimple starts when you dial from native dialer or when you launch the application).

When you set "Always available", close option make no sense : indeed, the application will automatically restart when network will change of state... and you'll say me "Hey there's a bug... the app constantly restart itself". By using this profile you tell the software that you want to receive your calls, and it will do it's best to be running when you can receive calls.

However, if you choose "Only for outgoing"... Close option will appear in the menu of CSipSimple dialer! Since here, it make sense to use this option, since the app will never restart itself.

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On 3G I don't always receive calls

• Your carrier network probably cut the UDP connection so that you can't receive calls anymore. To keep your connection alive you have to setup CSipSimple to be more aggressive.

To do so, transform your account into an Expert account (Edit account, press menu, "Choose wizard" and choose Expert). Edit again the account, there is now more settings to set. Try this :

• Re-Register (register timeout) : 184
• Keep Alive : 100
• According your device, you may also try to activate CPU lock.

Switch to ExpertSettingMode and in User interface settings activate "Use partial wake lock".

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I receive calls twice / Registration is done on the sip server twice

You SIP server probably don't support fully the RFC and the registration method used by CSipSimple by default is not suitable for your server. Fortunately, there is a way to configure your account to avoid that :

To do so, transform your account into an Expert account (Edit account, press menu, "Choose wizard" and choose Expert). Edit again the account, there is now more settings to set. Try this :

• Contact rewrite method : legacy
• (or if it doesn't work try to disable "Allow contact rewrite")

If your server actually doesn't support the latest RFC you may have to wait for about 15 min for the current registration to timeout.

In order to improve CSipSimple, if you are experimenting this issue and that this workaround help, you can inform us about that and tell us what is your SIP provider so that we could add a wizard with this enabled by default.

You can also notify your SIP provider that they are not fully RFC compliant (their server doesn't respect the normalization : and precising them that they do not respect RFC 3261 about contact rewrite method)

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I can't end calls

It's possible that your SIP server doesn't support actually TCP (in terms of SIP RFC compliance). Fortunately, there is a way to configure your account to force the use of UDP only :

To do so, transform your account into an Expert account (Edit account, press menu, "Choose wizard" and choose Expert). Edit again the account, there is now more settings to set. Try this :

• Transport : choose UDP is the list of transport.

In order to improve CSipSimple, if you are experimenting this issue and that this workaround help, you can inform us about that and tell us what is your SIP provider so that we could add a wizard with this enabled by default.

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Audio routing troubleshooting

If you are experimenting audio automatically routed through rear speaker instead of earpiece or other problem with audio driver on your phone, it's probably cause manufacturer implement a strange way its audio driver and android official API for routing audio.

Fortunately, there is some common behavior that can be managed by activating some "hacks" in CSipSimple.

To do so :

1. Install a dev version (HowToInstallDevVersion wiki page).
2. Switch to ExpertSettingMode (wiki page) global setting section.
3. Try to play with these settings (available in Media settings > ... (bottom of the list):
   • Use Mode API (try to use this one first and without activating others)
   • Use Routing API.
   • Use tone hack.
   • Use WebRTC implementation
   • Change the micro source (if related to the way micro records - big echo for example)
   • Change Audio Mode for SIP calls

Try each of these settings independently and then in combination.

If one or a combination of that modes helps... well good news for you, you have CSipSimple running correctly on your phone ;). You can then share with me which setting helps so that I can automatically turn on one of the correct hack by default for your device.

To do so, I need the exact device info (the infos announced by android to the application about the device). An easy way to collect these info is to send me logs using my embedded tool : see HowToCollectLogs.

If it doesn't work you can install other apps on the market such as "Device Info" and tell me what are the "Device" and "Product" values.

It's possible that none of these settings helps (for example it's currently a known issue on samsung moment). In this case we have to cross finger for your manufacturer or a custom ROM maker to fix that in the ROM directly. Maybe another hack could be found for your device too, but it requires devs skills and a phone to test on. If you are in this case, you can contact us so that we can give you what to test on source code to troubleshoot the problem.

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When screen goes off sound quality is bad

That's cause of the fact your device has special policy when screen goes off. There are a few well known reasons for it:

• Could be cause of the PSP behavior of the wifi card (see http://code.google.com/p/android/issues/detail?id=9781)
• Could be due to the fact you have setCPU that change CPU speed when screen goes off.
• Sometimes "Wi-Fi optimization" in the advanced settings for your Wi-Fi connection can cause packet loss and latency. Try disabling the setting and see if your connection improves.

Fortunately, there is an existing workaround in CSipSimple to prevent screen going off and so keeping up the good call quality. To activate this workaround if not automatically detected by CSipSimple (there is an autodetection done for the PSP behavior).

1. Switch to ExpertSettingMode (wiki page) global setting section.
2. In User interface section scroll to "Keep awake while in call" option and activate it

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In gingerbread with dialer integration I can't place GSM call anymore

Actually you can, you just don't know how :).

The first prompt you get is there cause both CSipSimple and the stock SIP application can intercept the outgoing call to treat it.

If you choose "Dialer", you actually choose the stock SIP application !! If you choose CSipSimple, you choose the standard process of telephony intent.

So just choose CSipSimple ! You'll see, next you'll have the CSipSimple chooser which allow to choose Mobile call.

If you don't want to be bothered anymore with the stock chooser that propose you between the stock 2.3 SIP application and CSipSimple, in the first popup (which CSipSimple don't and CAN'T manage), there is checkbox to remember your choice.

Activate it and choose CSipSimple. Then you'll not be bothered anymore with this chooser and benefit all powerful features of CSipSimple in terms of Filters and Rewriting rules.

In future release this will not be relevant anymore. You can try nightly builds if you are hurry to get something working well on gingerbread (there is also cool features for gingerbread included in latests nightlies)

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I did a mistake when I set up the voice mail number, how can I modify it?

• First of all, if you have to set up manually the voice mail number of your sip provider it means that the corresponding wizard for this sip provider has not yet the voice mail number automatically configured, so you should ask for that.
• If this is a non mainstream users provider or if you want to workaround quickly there is an easy solution :
  1. Go in accounts list
  2. Long press on the concerned account
  3. Choose wizard > Expert
  4. Click on the account row
  5. Scroll down to "Voice mail number", change it, and save
  6. Optionally you can revert to the previous wizard by reproducing point 1 to 3.

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I try to use ZRTP + SRTP, but ZRTP doesn't work

That's a known limitation. Due to the fact ZRTP is considered as a plugin for pjsip that already manages SRTP it's own way, you should never try to enable SRTP and ZRTP at the same time.

In fact if you do so, pjsip will ignore ZRTP and use SRTP as media adapter.

It's recommanded to use ZRTP that includes features of SRTP if the remote side is known to also support ZRTP. If your remote side is a sip server that only use SRTP and announce it with SIP mechanism, you'll hove to choose SRTP however.

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Cryptography notice

This distribution includes cryptographic software. The country in which you currently reside may have restrictions on the import, possession, use, and/or re-export to another country, of encryption software. BEFORE using any encryption software, please check your country's laws, regulations and policies concerning the import, possession, or use, and re-export of encryption software, to see if this is permitted. See <http://www.wassenaar.org/> for more information.

The U.S. Government Department of Commerce, Bureau of Industry and Security (BIS), has classified this software as Export Commodity Control Number (ECCN) 5D002.C.1, which includes information security software using or performing cryptographic functions with asymmetric algorithms. The form and manner of this CSipSimple distribution makes it eligible for export under the License Exception ENC Technology Software Unrestricted (TSU) exception (see the BIS Export Administration Regulations, Section 740.13) for both object code and source code.

[Anonymous]

Originally posted by: j.soo...@gmail.com

Incoming voice is played via speaker instead of earpiece causing the other party to hear themselves speak. This forces me to use headphones. How can I make csipsimple use the earpiece for playback like standard calls?

[Anonymous]

Originally posted by: r3gis...@gmail.com

@soong1 : you should try one last dev version (see HowToInstallDevVersion). I've recently this kind of problem on some devices. If it doesn't not help, you can use the issue section to report a detailed issue (specially what device you are using).

[Anonymous]

Originally posted by: j.soo...@gmail.com

I'm already using the latest dev version and it doesn't help. I'm running on a Samsung Intercept with OS v2.1. I'll try to write an issue report.

[Anonymous]

Originally posted by: j.soo...@gmail.com

After filling out the 'new issue' form the submit issue button is greyed out and not clickable for me.

[Anonymous]

Originally posted by: PureLone...@googlemail.com

Hi there

I have the app set to be selectable each time I choose to dial - But I would rather that it didn't launch the application unless I choose it...

Is this possible? I have a very specific SIP setup and only want it to launch when I specifically tell it to.

Thanks

[Anonymous]

Originally posted by: radu_ram...@yahoo.com

How can I choose what codec is used ? For instance for me it only works with 8khz pcm .( pressing info during call I can see this) When I enter the menu there are 4 codecs available and some unavailable. But is there any way I can convince the softphone to use GSM instead o 8khz pcm as I guess GSM is more bandwidth efficient over EDGE network.

On wifi it works awesome regardles the settings or codec.

[Anonymous]

Originally posted by: r3gis...@gmail.com

@LoneWolf? : In this case you should probably choose to disable android integration and use CSipSimple dialer when you want specifically to dial out with SIP. You can also play with rewriting/filtering rules (see the corresponding section above).

@Radu : To change codec orders : in Settings > Media > codec drag and drop using the little handle at the left of the screen. To disable a codec : long press and deactivate/activate codec. Codec order here are just what CSipSimple advise. According to the settings (and supported codecs) of your remote party negociation can finally always lead to the same choose. To force the use of a codec you can try to disable all others. However be careful if you do that : if other party doesn't support the only codec you have left activated, call will directly hang up.

[Anonymous]

Originally posted by: j.soo...@gmail.com

I would just like to update that I've been trying each new dev release and using the various combination of commands like "Use routing API" and "Use Mode audio API" but the problem of incoming audio going to speakers instead of earpiece persists. Thanks for you consideration.

[Anonymous]

Originally posted by: ladams0...@gmail.com

@j.soong1 and @r3gis.3R - Same exact issue with Samsung Moment (older brother to Samsung Intercept). Use Routing API and Mode audio API combinations ineffective. Using build 0.00-16 r410

[Anonymous]

Originally posted by: dr.k...@gmail.com

ditto @j.soong1 @ladams0000: audio only via external speaker, silent handset when speaker is disabled.

[Anonymous]

Originally posted by: cristina...@gmail.com

Hello,

Is there any way of looking at the logs? I can send message to another devices but not received messages... instead of receiving the message the app is closed.

I've try to debug the code using the Android eclipse emulator but the welcome page doesn't allow me to use the app because is allways trying to download the info.

[Anonymous]

Originally posted by: dengx2....@gmail.com

Hello. I pretty much thanks for your contribution.

I experienced so long delay about 500 ms. It means one way delay.

I think it is not a network problem. Is there any way to solve this problem?

[Anonymous]

Originally posted by: ken.laberteaux

What is Tone hack?

[Anonymous]

Originally posted by: r3gis...@gmail.com

See  issue 371  For expert settings most of the time it's a good idea to do a quick search over all issues in issue list (not only opened issues) ;)

[Anonymous]

Originally posted by: iiordanov@gmail.com

Hi guys,

I've written a guide on how to combine a SIP client like CSipSimple with Google Voice and the Gizmo5 SIP service to make free calls to USA and Canada over WiFi? or data without using your voice carrier at all here:

http://iiordanov.blogspot.com/2009/07/sipdroid-gv-guava.html

This works from anywhere in the world as long as you have or can create a Google Voice and Gizmo5 account.

[Anonymous]

Originally posted by: pindi...@gmail.com

hi, just started using it first time with a betamax voip provider named powervoip, it seems to be working but the call quality was horrible, both of the sides were not able to listen to each other.....(the same voip account works ok on my pc though). Further when i tried to put headphones during a call....it crashed i think and restarted my wildfire...(froyo). Thanks..waiting for the improvements....

[Anonymous]

Originally posted by: pindi...@gmail.com

in addition to the last comment, I just tried Stun server settings and that worked great for me and now the voice quality issue is gone.....:)

[Anonymous]

Originally posted by: nilcasd...@gmail.com

I have in my Sony Ericsson Xperia X10 mini Pro the same earpiece/speaker problem the Samsungs have. Can you help me ?

[Anonymous]

Originally posted by: r3gis...@gmail.com

@nilcasdias : is your X10 mini pro running android 1.6 or 2.1? (On my X10 mini it works correctly but android version is up to date : 2.1, previously sony had bugs with their audio driver on X10, X10 mini, mini pro and X8 that they have fixed AFAIK in android 2.1 update).

[Anonymous]

Originally posted by: gregmaw...@gmail.com

@nilcasdias - I have an X10 mini and upgraded to 2.1 a couple of days ago. I have loaded CSip today and so far only made 1 call. The sound quality was generally very good. Very clear and understandable and was using the inbuilt speaker/mic with no problems after I enabled STUN. There was a little choppiness in the other persons voice but otherwise I'm very happy so far.

Thanks very much to the devs of this app!!

[Anonymous]

Originally posted by: gauvi...@gmail.com

Bad gateway on 3g. Sipdroid connects without problem. Bug or settings?

[Anonymous]

Originally posted by: r3gis...@gmail.com

@gauvin : it's a setting issue. Can you open an issue on issue list on issue tab of this website ? And precise your sip provider? I'll explain you how to configure and we will create a new wizard for your provider so that other users will not experiment the same problem.

[Anonymous]

Originally posted by: ongr...@gmail.com

I get an voice mail icon and i dont have any .. anyway to disable it? its happening only with the rc versions scened Q is it possible to disable the "slide" graphics ?

[Anonymous]

Originally posted by: r3gis...@gmail.com

@ongrass : for voice mail for now no option to disable voice mail integration. This new feature is based on what is sent by your sip provider : if csipsimple show a voice mail notification it mean that the sip provider announce you have an unread voice mail. But I'd be interested by logs if you can collect and send me some logs I'd be interested. For slide graphics : there is an option : see ExpertSettingMode wiki page to switch to expert mode and then in User interface choose "don't use slider for call answering" option.

[Anonymous]

Originally posted by: azeem.za...@gmail.com

Cannot connect with voipbuster at my HTC Desire HD. www.voipbuster.com This one of the service provider from Betamax Germany. works OK with voipalot.com and siptraffic.com