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FAQ
Summary
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). Before startingIf you don't know what is SIP and what this program does : WhatIsSIP The other party can hear me but I can not hear them
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"
In this case, try to disable Bluetooth option of your phone (even if nothing is paired).
There is a vibrate icon in the notification bar while in callThat's absolutely normal. In order to avoid your ear to explode if there is an incoming GSM call, I put the ringer mode to silent (that's the mean of the icon in the notification bar). When call will end, ringer state will be restored to previous value. There is no contact listJust 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. When integrated to android, I don't want some number to be handled by SIPI want to automatically add prefix/rewrite some numbersThere 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 :
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 :
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. On 3G I don't always receive calls
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 :
Switch to ExpertSettingMode and in User interface settings activate "Use partial wake lock". I receive calls twice / Registration is done on the sip server twiceYou 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 :
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) I can't end callsIt'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 :
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. Audio routing troubleshootingIf 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 :
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. When screen goes off sound quality is badThat's cause of the fact your device has special policy when screen goes off. There is two well known reason for that :
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).
In gingerbread with dialer integration I can't place GSM call anymoreActually 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) 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 workThat'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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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?
@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).
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.
After filling out the 'new issue' form the submit issue button is greyed out and not clickable for me.
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
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.
@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.
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.
@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
ditto @j.soong1 @ladams0000: audio only via external speaker, silent handset when speaker is disabled.
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.
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?
What is Tone hack?
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) ;)
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.
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....
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.....:)
I have in my Sony Ericsson Xperia X10 mini Pro the same earpiece/speaker problem the Samsungs have. Can you help me ?
@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).
@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!!
Bad gateway on 3g. Sipdroid connects without problem. Bug or settings?
@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.
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 ?
@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.
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
Will not instal on G-Tablet (Vegan 5.1 ROM) 3CX working fine :(
@vaiovill : try new distrib mode : http://nightlies.csipsimple.com/trunk/
Will do, THANKS!
This time it installed fine and it only force closed on first open After that it works fine :) Thanks again !
What rfc and spec is SIP code complaint to ?
@sunild : have a look to pjsip home page. CSipSimple is based on pjsip stack so each features of CSipSimple comes from features of pjsip ;)
I am looking to add additional custom headers using csipsimple API, but couldn't figure out how does msg_data and hdr_list of pjsip maps to csipsimple's Java functions, and how can I use them to add custom headers, e.g., "X-MyHeader?: Some Value" in the SIP message. The pjsip itself has ways to add custom headers, e.g., http://svn.pjsip.org/repos/pjproject/trunk/pjsip-apps/src/pjsua/pjsua_app.c shows how to add Warning header in function "call_timeout_callback".
Never mind the previous comment. Looks like I will need to add additional functions in pjsua_wrap.cpp and pjsuaJNI.java to support header/message data operations.
I have been trying to use csipsimple with VOIP service whistlephone.com... whistlephone registers successfully...but on making a call, i can jst hear the other side for 2-3 sec...and then no sound at all whistlephone uses these codes- iLBC (select platforms), G.711 u-law and G.711 a-law please help
Cannot connect with nonoh.net at my HTC Desire . www.nonoh.com This one of the service provider. works OK with sipnet.ru
nonoh works on my HTC desire: Make an expert account: account name :nonoh account id:'internationalphonenumber'<sip:'internationalphonenumber'@sip.nonoh.net> reg URI: sip:sip.nonoh.net username:nonoh username data password: nonoh password everything else default.
HTC Desire: clean install. SIP calls working well w/o filters. I could set up any filter correctly, but nothing of them works at all. Wiki checked, expert mode checked. Anyway, its a good app ;)
Thanks for such a wonderful app! I have Evo froyo. Only issue for me is no Bluetooteh support. Using both stun & ice. Audio is great.
Google Nexus S: I can hear the third party but they do not hear me. Any idea what's wrong?
hey guys.. any idea on how to setup nymgo with csipSimple on android.
I used:
username (without ata.nymgo.com) password XXX server ata.nymgo.com
In settings i also turned on STUN and used stun.nymgo.com as stun server address. ICE is also turned ON.
Account is getting regestered but I am unable to make a call on 111(diagnose number), it drops down without even ringing.
I am using wifi network.. latest version of csip.. android 2.2.. Dell Streak
Hi. I hope this is the right place to ask this question. Can you please give me a clue about the way CSipSimple handles the call status? I saw in SipCallSession? that error messages like 380 (alternative service) are defined in the list of possible StatusCode?, but it looks like the notification about those statuses are never received. I have the impression that CSipSimple does not get those notifications from pjsip. Is that correct? If not where do you handle error code like 3xx, 4xx. Thank-you
Status code is never shown to the user. Mainstream users will never know what does a status code mean. The comment sent by the server is shown however. If you want to read exactly how CSipSimple ui treat what comes from CSipSimple service (and comes from pjsip), read this class : http://code.google.com/p/csipsimple/source/browse/trunk/CSipSimple/src/com/csipsimple/utils/AccountListUtils.java
Thank-you for your prompt replay. And sorry for my long question. Even if you don't need to show the status code to the user it should be possible to handle cases like the 380. Is this error code sent from the pjsip lib to csipsimple? Another thing that I noticed is that in SipCallSession? the callState stores the pjsip_inv_state from pjsip (remapped in the InvState? constants) and not the StatusCode?. What about the StatusCode?? That status code is used in the class you suggested me to check: AccountListUtils?. But that class is about the profile state and not the callstate. I saw that SipService? has the getCalls method. I was expecting to be able to get the status of all the calls. What I saw is that getCalls returns an array of SipCallSession? and as I wrote that class stores the InvState? and not the StatusCode?.
So my doubt is: Are the 3xx received by csipsimple from the pjsip lib? Where and How?
Thank-you again
Ok, will become more technical so do not hesitate to reply me by mail. (the FAQ is for mainstream users not for developers ;) - another place would be the developers mailing list).
About the sipcallsession, (my bad I thought you were talking about the sipprofile), what you are looking for is the getLastStatusCode.
But place a mail on the developer mailing list. I don't know what you are trying to do, but I could probably be helpful. The java part is maybe not the right place to put treatment of the return codes of the native library. (Also, just as reminder, the app is released under GPL license ;) ).
I have a Samsung Prevail .... I read the above for troubleshooting audio routing to the back speaker away from Bluetooth ... Any specific suggestions for this phone ... I wasn't able to find the mode settings menu that you spoke of. TY. CMJ. cmjllc@gmail.com
Is it possible for you to try a remote on the phone or I can send it to you .... ?
Is there a way for you to remotely work on phone or for me to send the phone to you to take care of audio problem of the calls being routed to back speaker (away from Bluetooth)(Samsung Prevail)?
I can't send text messages. I always have to force close. That is quit annoying
@moment... : can you try to collect logs (see HowToCollectLogs wiki page). I'll maybe find some interesting clue to fix the bug :). thx
I can't get the zoom to make calls with the app are their certain settings I need to put on. Currently using 3.1os honeycomb wifi only xoom
When I try dialing a number it keeps telling me to add it to the contacts and even when I do it doesn't allow me to make the all not sure if its a tablet issue....
If you have a problem with the speakerphone not turning on, but it used to work, go to Expert mode > Media > Use WebRTC Implementation. It still doesn't fix echo over speakerphone, but with mute on it is somewhat useful. D2G.
I installed on my Samsung Charge and cannot make calls. I am using whistlephone service and it shows REGISTERED. I CAN receive calls and the sound quality is great over 3G. When I try to make a call I hear ringing but I never hear the WhistlePhone? ad or does the phone I am calling ever ring. I am using Verizon in the USA. I have tried changing many of the settings but nothing helps. Can you shed some light as to what I need to do to make calls too.
Update, inbound calls do not always come through.
Hi guys,
I've written a guide on how to combine a SIP client like CSipSimple with Google Voice and Sipgate or IPKall 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. Also, if you only want to get a US number and receive calls for free, you can follow my other two guides to get a Sipgate or IPKall number:
http://iiordanov.blogspot.com/2011/06/how-to-get-free-sipgate-account-with-us.html
or
http://iiordanov.blogspot.com/2011/06/how-to-get-free-ipkall-us-number-did.html
Cheers!
Hi guys!
I´am using csipsimple and no problems most everithing fine.But when I try to make the registration with a Betamax in Portugal with Meo network using a thomson router I can´t but with Mydivert I can. This problem is only with Betamax clone and Meo network.Can You help me please.
Hi r3gis, Thanks for the free app. I've got a T-mobile G2x, installed the CSipSimple and am able to receive and make calls through SipSorcery?+Google Voice+IPComms DID over both WiFi? and t-mobile's data.
There is one problem: when I bridge CSipSimple and a Linksys VoIP adapter directly (through SipSorcery?), everything is fine when CSipSimple is on WiFi?, but not fine on t-mobile data. No matter which side initiates the call, ringing is fine, CSipSimple can hear Linksys box but Linksys can not hear CSipSimple.
i have enable Native client integration on CSIP. However, when I call a phone I don't see CSIP in the list (I do see the Regular Dialer and Viber). I'm using it on Samsung S2 with android 2.3.5, any suggestions?
Choose "regular dialer" it will show up the csipsimple dialer integration !!!
CSipSimple integrates a different way than skype and viber that does something not really user friendly. The way done by csipsimple allow you to use filtering/rewriting rules and does not always show, but only when necessary. There is already a plugin for skype so that csipsimple integration popup show the skype choose. We should add one for viber to if they also integrate this crappy way.
On the first popup, choose "dialer" and check "remember my choice". It will always go through csipsimple integration which also allow you to create filtering rules so that the popup is only shown when you actually want to show it. And not always like it's done by other apps.
Thank you very much, it worked
i have a NAT problem when i use CsipSimple? under normal router i get the ip address is 192.168.1.101 but i always see the registered info in the proxy is router's WAN IP just like this sip:1925XXXXXXX@112.205.213.81:36897 not sip:1925XXXXXXX@192.168.101.:36897 so i always got one way voice , where i can set this issue ?
@simon : try the point "I receive calls twice / Registration is done on the sip server twice". With disable the feature.
how do i send messages from the dialer
Hi I have been using csipsimple for a while now and I have found good sound settings. I am now very satisfied with this application. I am now trying to install a handsfree system in my car. I have installed a mic to use along with my car speakers. My problem is accoustic echo. The other person is hearing his own voice. Since I am not very familiar with echo cancelation, I was hoping to get informations on any settings I could try to make the accoustic echo dissapear. Thank you for this great application. The best I tried yet...
Hi, I am trying to get my BT earpiece to work with CSip and my Samsung Galaxy tab, and I see there is a sound fix provided for Galaxys under "expert" settings, however, this does not work for me. I did get momentary/intermittent sound through the earpiece by playing with the "Audio mode for SIP calls" setting. Do you have any further recommendations for Galaxy owners? Thanks for helping and thanks too for your great app!
Hi. How can we find out the kind of upgrade that have been made from one nightly builds to the other?
I have setup the CSipSimple on Galaxy SL for using with Nymgo, and it's working perfectly now after some of the configurations such as UDP/Galaxy S Hack etc. Here is the link to it.
http://technology-shettyprasad.blogspot.com/2011/12/configuring-csipsimple-sip-client-on.html
and BTW, this FAQ is also very useful in sorting out many issues.
I have a problem when using WiFi? at home. When I'm on a 3G network CSipSimple works properly. When I am connecting through my router on WiFi? the party I call cannot hear audio, but I can hear them. So I set the UDP Port in CSipSimple to 5060 and also forwarded that port in my router to my Phone's IP which is assigned through DHCP. But that did not change anything. Obviously the problem lay somewhere with the configuration I have in my router, but short of the Port fowarding, I'm not sure what to try next?
I have one ringtone for the 'mobile' number and then another for all the SIP accounts.
Is there any way to have a different ring tone for different SIP accounts? Then I can hear from a distance if it is a more important number that is calling !
I am trying to register my magic jacks info on my csipsimple application and I have all my info filled in but am getting "error while registering Bad Gateway" ..
Csipsimple doesnt work whith my headset bluetooth... well, It works but i cant hear anything. How to resolve it? Thanks
I'm using CSipSimple on my "HTC Desire HD" (Android 2.3.5) and also on my "Asus Transformer Prime" (Android 4.0.x) Allways when I missed a call, a sounds starts, which I can't stop. I have to shutdown the mobile or the table. I couldn't find any other way to stop playing the alarm! Can anybody tell me why or how I can solve this problem?
Thank you in advance Gerrit
Bluetooth issue: Atrix(stock rom) + Jawbone ICON, no sound. when the bluetooth radio is on, even if no earset is paired or on speaker mode, sound cannot be heard whatsoever. For this case, Sipdroid works just fine.
on r1306(latest) and r1297?
Samsung Galaxy Mini with CSIPSIMPLE. Will not answer call. Any Ideas ?
I am having trouble with volume on LG p990. Using Engin Voip Australia codec G729.
Sorry, do changes, described in "Audio routing troubleshooting", have any affect on other apps? Standard phone, for example.
Do anything please with working with bluetooth!
Normally no. Unless the manufacturer did very very buggy android code for their device. Besides normally everything done by csipsimple to change audio routing is restored once call ends. Anyway, in case of very buggy android ROM from manufacturer, a reboot will restore everything in correct state.
BT suddenly begun to work in CSipSimple. But when CSipSimple is active, BT doesn't work on incoming calls on standard phone.