No audio (in and out)

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What steps will reproduce the problem?
1.Make a call
2.Recipient pick up the phone
3.There is no audio in/out

What is the expected output? What do you see instead?
Expected audio between both parties.
Got no audio and no dial tone.

What version of the product are you using? On what operating system?
Motorola Droid, 2.2, sip provider: voipbuster

Please provide any additional information below.
To get audio I have to press and release soft HOLD button at the upper left 
corner of the app. What interesting is that it happened only when i make 
domestic call... Everything is working fine when it is international call.

Original issue reported on code.google.com by yaleks...@gmail.com on 25 Aug 2010 at 5:57

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Really interesting.

I'll need a little bit more infos :
1 - Is bluetooth enabled? (I mean not is there a BT handset, just if BT is 
activated in android)

2 - Is that possible for you to provide me some logs :
    * Go to option > settings > Ui> log level (last entry) and set it to 4
    * Reproduce a failing use case
    * Use a logcat app (such as alogcat available on the market) to send me your logs.

3 - another thing you can try is setting the sip proxy field (see instructions 
on issue 164 / comment 3 /problem 4

Original comment by r3gis...@gmail.com on 25 Aug 2010 at 12:53

• Changed state: Accepted

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I noticed if you don't use international extension even for domestic calls sip 
wont work.

Original comment by alessand...@gmail.com on 25 Aug 2010 at 3:15

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Hi,
I've sent you my phone's log by the aLogcat.

1. BT was disabled.
2. Set just like you said.
3. Tried that with no success.

Thank you

Original comment by yaleks...@gmail.com on 25 Aug 2010 at 3:38

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@yaleksbox : didn't receive your logs. My email address is r3gis.3r at 
gmail.com 

Original comment by r3gis...@gmail.com on 25 Aug 2010 at 9:11

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Anything? News?

Original comment by yaleks...@gmail.com on 27 Aug 2010 at 6:10

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According to the logs there is something that finish the call before it is 
completely established.

On the log you sent to me, is the in call screen reach the green state 
(confirmed call / talking)? Or did it directly go from gray (ringing) to red 
(hangup)?

Seems the remote endpoint either hangup or decline your call (can be also the 
sip server that act like that).
* If you decline the call for the log, please try to take the call and provide 
me the log with an established communication (a few seconds will be enough).
* If you always reproduce this use case, the problem is not that there is no 
audio, but that the call is never established. As it seems to well negociate 
the codec (it use ilbc - not the best choice to start but should be ok even if 
choppy - you can try to disable it in the settings to start with a better and 
reliable codec). I think more that it's your server that doesn't understand the 
contact you are trying to call. Are you sure your sip server understand the "+" 
char? some configuration need it to be replaced by 00. 

Original comment by r3gis...@gmail.com on 27 Aug 2010 at 8:07

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@yaleksbox make sure you can dial with some otyher phone using the string you 
are using. I don't know about voipbuster but my provider Callcentric cannot 
understand the '+'. You must dial domestic (zone 1) calls 1aaannnnnnn and 
international calls (other zones) 011ccnnnnnnn... 

Original comment by dc3de...@gmail.com on 28 Aug 2010 at 12:46

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I've noticed bluetooth has not worked since there were some changes to 
bluetooth with versions 24 and later, so I am using 23.  I have to switch bt on 
and off 2 or 3 times but it does work with v12-23.  

Original comment by tdbj...@gmail.com on 3 Sep 2010 at 5:34

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I have the same problem on HTC Desire 2.2
It rings but when recipient picks up, no audio In or Out.
If I press the Pause button (pause and unpause), then it works well.
A bit of echo tho...

Original comment by gleveill...@gmail.com on 16 Oct 2010 at 8:01

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Could you try to activate stun (settings > media > activate stun).

Original comment by r3gis...@gmail.com on 16 Oct 2010 at 8:06

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Any news with latest version (0.00-15 available on the market)?

Original comment by r3gis...@gmail.com on 17 Oct 2010 at 11:15

• Changed title: No audio (in and out)

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I have similar problem (Samsung Galaxy i5800). No sound on call. Also, when a 
call is in progress, the picture meaning silent mode is seen in upper side of 
the screen.

Original comment by ser...@gmail.com on 1 Nov 2010 at 10:48

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Sorry, forgot to give more info. Android v.2.1, cSipSimple all versions 
available for downloading, SIP provider Callcentric.

Original comment by ser...@gmail.com on 1 Nov 2010 at 10:56

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HEllo, I earlier posted a problem regarding no audio when making
outgoing calls, now I have also trod some suggestions available here,

Soft pressing the hold button and them resuming gives sound, other writer
There is no sound.
Galaxy s, froyo ,csipsimple,15-17  trying on sipgate
Bluetooth disabled / switched off from phone.

Original comment by aamir...@gmail.com on 4 Dec 2010 at 3:32

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Tried with Motorolla Mailstone (2.1-upgrade-1) & HTC Desire A18181 (2.2)

on both cases I do not hear anything during the call. Media streams are 
correct. On log I see error message "AudioMgr Error:Invalid output format flag; 
disabling PostProcessing"

I have recorded call - file is correct sound is present.

Samsyng Galaxy I9000 works fine.

I am ready to send any logs and make any examples

Original comment by dku...@gmail.com on 14 Jan 2011 at 3:37

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@dku : your devices (desire and milstone) are probably affected by the PSP 
problem.

For HTC desire, it's highly possible cause it's already known that HTC has PSP 
behavior when screen goes off. 
In latest dev version (http://nightlies.csipsimple.com/trunk/) it should be 
correctly auto-detected now and automatically activate the workaround against 
PSP problem.

On Milestone I'm less sure. It's maybe PSP. You can try to activate the PSP 
workaround manually.
Activate ExpertSettingMode (wiki page => for global settings), and in User 
interface > activate Keep awake while in call.

But could also be some routing issue. If so maybe worth to try what is listed 
here :
http://code.google.com/p/csipsimple/wiki/FAQ#Audio_routing_troubleshooting 
Audio routing troubleshooting section

Let me know how it goes.

Original comment by r3gis...@gmail.com on 14 Jan 2011 at 6:23

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1. I have made tests between Sony Ericsson Xperia X8 & x10 
There are better then on previous tests (HTC & mailstone ) but still not good 
enough. Will try to make additional tests later.
2. On my point of view there is not linked with screen off.
3. Tested under Samsung I550 - works fine.
4. For HTC & Mailstone tried to play with setting from trouble shouting list. 
No changes. 

How do you think Can it be linked with wrong sound device driver? Possible 
problem should be sorted out on PJSIP sound device level?

Original comment by dku...@gmail.com on 18 Jan 2011 at 1:35

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It seems to be a NAT problem.

Original comment by joze.rov...@gmail.com on 21 Jan 2011 at 3:01

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I have an HTC Desire running froyo, Csipsimple works only with PBXES.org as 
voip provider, my regular provider is freephoneline.ca I couldn't make work as 
always not sound in or outbound, nimbuzz on the other hand works well very good 
sound, the only problem is that gets disconnected from wifi after a while, so 
inbound calls don't get through. Help Please, I want this to work!

Original comment by enciso.d...@gmail.com on 20 Feb 2011 at 12:51

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Issue 760 has been merged into this issue.

Original comment by r3gis...@gmail.com on 3 Mar 2011 at 9:19

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I use csipsimple with sipgate.  The default STUN server for sipgate is used and 
activated, and so is ICE; however, I lose my incoming audio most of the time, 
although I do briefly get it back here and there. The loss is probably due to 
changing between wifi and GSM. 

I'd like to try changing the STUN server to see if that helps me out at all. 
Would any other STUN server work with sipgate?

(I've got an LG Optimus V.)

Would it be worth just considering a different voip provider?
I think I could take any that provide free inbound calls,
is that the standard - the great majority of them?

Original comment by nate.ka...@gmail.com on 16 May 2011 at 2:39

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I use CSIPSIMPLe on Cyanogenmod 7.0.3 with my HTC Desire connecting to my own 
Asterisk server and I have the exact same issue: although I can register w/o 
much probblems, I have no incoming sound at all.
I tried playing with the codecs (I usually prefer using ulaw) but it didn't 
change anything.
Then I Also tried to set up stun.3cx.com as a stun server but it didn't resolve 
the issue.

Original comment by teho...@gmail.com on 25 May 2011 at 12:18

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@teho : you could maybe try to follow these instructions about routing :
http://code.google.com/p/csipsimple/wiki/FAQ?wl=en#Audio_routing_troubleshooting

I'm not sure it will help, but since CM7 on HTC desire audio driver may not 
integrate the new API for sip calls, you should try to revert to default modes 
(instead of the one I set when I detect Gingerbread cause I assume all 
manufacturer did things to support the new audio modes) :
Micro source : select default instead of communication
Mode for sip calls : select normal instead of in_communication.

Original comment by r3gis...@gmail.com on 25 May 2011 at 1:24

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Tried changing all those options, tried all of the other choices in both menus, 
tried changing the API Modes, the Galaxy hack and everything else I could think 
of (codec rates etc.), nothing would do...
I also reverted to using a local authentication on the same network as the SIP 
server (avoiding STUN problems), it didn't change anything.
According to my asterisk server logs, everything looks fine, sounds are played 
correctly and there's no visible handshake problem.

That's really a pity, that softphone really looks incredible, GPL code, loads 
of options, recording and all I could ever dream of... Damn.
CM 7.0.3 is Android 2.3.3.

Original comment by teho...@gmail.com on 25 May 2011 at 11:45

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I am having the same issue with a Motorola Milestone. Some outgoing numbers 
work fine but some have no audio in either direction. It seems to be a problem 
with the integration with the Android Dialer as it does not seem to happen when 
making calls directly from CSipSimple. I have one number that fails everytime 
from the dialer, which happens to be a Google Voice number. It does work if I 
make the call through CSipSimple. I am using Android 2.2.1 and FreePhoneLine.

Original comment by ned...@gmail.com on 7 Jun 2011 at 6:14

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Usually when it does not work on one way only or on one kind of number only the 
problem is related to codecs. Some sip providers announce to support codecs, 
but actually does not gateway media. So usually disabling codecs that are not 
correctly supported by the sip provider solve the problem

Original comment by r3gis...@gmail.com on 17 Jun 2011 at 10:08

• Changed state: Obsolete