No sound on Desire 2.1

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While in call impossible to get sound from speaker/earpiece. Micro works 
properly.

Original issue reported on code.google.com by r3gis...@gmail.com on 10 Sep 2010 at 10:20

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Is there a custom application that modify sound settings?

Original comment by r3gis...@gmail.com on 10 Sep 2010 at 10:21

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No, I don't see other than the default music player.

Original comment by enciso.d...@gmail.com on 10 Sep 2010 at 10:24

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Ok, strange.
I get a lot of
D/AudioFlinger(   57): AudioFlinger::MixerThread::prepareTracks_l(), 
track->isMuted(0), track->type(0), mStreamTypes[track->type()].mute(0), 
mStreamTypes[track->type()].volume(1.000000)

Since it is from the android system that's really strange.
Is there an handset (Bluetooth / Headset) wired on your phone? 

I assume it is an official ROM regarding build number but it seems to be really 
verbose. Is there some special things about your ROM or configuration ?

Original comment by r3gis...@gmail.com on 10 Sep 2010 at 10:36

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1st quesion, yes, I have tries with bluetooth headset, wired headset (htc and 
3rd party) and also without any headset just the phone's earpiece.

2nd question, Yes it is official, HTC Telus comm. but unlocked (HTC provided 
the unlock keys) I have not root or done anything strange to my phone. Just 
waiting patienly for Telus upgrade to 2.2

Original comment by enciso.d...@gmail.com on 10 Sep 2010 at 10:45

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Ok, so it's a standard conf. Indeed that's really strange it doesn't work (all 
the more so as htc usually do things cleanly with their ROM) :S... So probably 
something to see with my app.

I'll try to read a little bit more the android source code to see why we get 
this log from the android audio stack.
Next Monday, I'll ask a colleague if I can test on his Desire 2.1. (It's an 
European version of the ROM but should behaves the same).

If I see something in the android code that there is something that could be 
interesting to test before Monday, I'll ask you on this thread.

Thanks for the report !

Original comment by r3gis...@gmail.com on 10 Sep 2010 at 10:57

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Sure no problem thank you.

Original comment by enciso.d...@gmail.com on 10 Sep 2010 at 11:04

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Hi There,
something really strange happened, audio is working now, no idea how, i didn't 
change anything but now is working, the problem now is that is working only in 
WiFi not on cellular network and I plan to use it mainly on cellurar network 
(3G) wich I now my carrier allows. Any thoughts?

Original comment by enciso.d...@gmail.com on 16 Sep 2010 at 3:01

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Ok, that confirms my tests with the phone of my collegue (worked properly over 
wifi - but I didn't test over 3G since our provider doesn't allow it).
Most of the time in such case, can be due to the fact something in the SIP 
configuration doesn't allow to pass through the NAT of your 3G carrier.
Have you a stun server configured? In many case it can help with this kind of 
issues. 

To enable stun server : Settings > Media settings > Tick enable stun and fill 
stun server with a stun server. If your sip provider has a stun server use it, 
else you can try stun.ekiga.net for example.

Original comment by r3gis...@gmail.com on 16 Sep 2010 at 8:00

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Sorry Mate, I could not play with the settings anymore, I just could not make 
it work, it worked for a few hours and then it didn't work anymore. no audio 
while in call, and lately, it drops the call a few seconds after is connected, 
i tried all versions of the program, and so far none worked, and that is a 
shame, because i love the way your app integrates with android, now I have to 
use  3CX for android which I hate but gets the job done without any trouble.    

Original comment by enciso.d...@gmail.com on 16 Sep 2010 at 12:31

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That's really strange, all the more so as 3CX is based on the same sip stack 
than the one I use. (But unfortunately, they don't open their source).

Well stay tuned on the dev on csipsimple. As I don't reproduce your issue, it's 
hard to tell when I'll be able to solve it. But with chance, as CSipSimple is 
opensource, maybe in the future it will really improves (and remember, 
CSipSimple is still in its alpha phase ;) )

Original comment by r3gis...@gmail.com on 16 Sep 2010 at 6:24

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Just out of curiosity whouldn't the GPL license of pjsip require 3CX Phone to 
be released as open source if that is what they are using?

Original comment by michael....@gmail.com on 16 Sep 2010 at 10:31

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I believe that's the rule, I am doing a little bit of research and apparently 
they use a combination of open source and proprietary but if they use open 
source they need to release it. Their system works in my HTC Desire with no 
problems, their UI is horrible,  and it doesn't integrates very well with 
android. I wish it were like your app mate, I will keep watching your 
development.

Original comment by enciso.d...@gmail.com on 16 Sep 2010 at 11:08

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I've asked 3cx devs about license issue.but that's ok since they buy the dual 
license of pjsip. So their product is ok regarding the law.
Not really cool regarding community but that's their choice, and hopefully the 
opensource project will become better. Your efforts to reports and help to 
debug csipsimple will make it better ;).

Original comment by r3gis...@gmail.com on 17 Sep 2010 at 3:41

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A new development, now I can make calls on WiFi with perfect sound both sides 
but when I change to GSM, no sound the other person can hear me but I can hear.

Original comment by enciso.d...@gmail.com on 23 Sep 2010 at 9:04

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If you change while in wifi communication, that's normal for now.
Hand over is not yet supported.

Original comment by r3gis...@gmail.com on 25 Sep 2010 at 6:17

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HI, I have the same problem on an LG GW620r. It is running Android 1.5. Several 
SIP phones give me very low audio, your software gives much better volume, but 
sometimes mutes the phone, and then I have to exit the program to reset it. Too 
bad, because I like it a lot, but can't really use it.

Original comment by davidp...@gmail.com on 12 Oct 2010 at 2:15

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Could be a NAT problem :
I've just added a question in the FAQ :
http://code.google.com/p/csipsimple/wiki/FAQ#The_other_party_can_hear_me_but_I_c
an_not_hear_them

Could you try to follow the instructions and say me if it helps. On Desire 
quietly sure that's the problem : 3cx automatically add their stun server 
stun.3cx.com to get ride of NAT issues. As CSipSimple is not linked to any 
provider, I don't add a STUN server by default. I don't know the term of use of 
each stun providers so that I could use by default one stun service in 
csipsimple.

On LG : not sure that could solve the issue, if you constantly use the same 
connection, the problem is maybe somewhere else (but worth to try stun). So 
davidpnew if you reproduce with stun enabled, could you open another bug for LG 
GW620 support?

Original comment by r3gis...@gmail.com on 12 Oct 2010 at 6:38

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I solved the problem on the GW620r  - if I modify the audio stream and speaker 
volume settings within the program, it still shows the silent mode indicator in 
the status bar while the call is in progress, but the signal becomes audible.

Original comment by davidp...@gmail.com on 12 Oct 2010 at 1:04

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@davidpnew : don't be worried about the little icon in the status bar. 
It is in order to avoid your ear to be exploded by an incoming GSM call while 
you are in a sip call.
I set the ringer mode to silent... But will not affect the audio streams (nor 
mic nor speaker ;)) !

Original comment by r3gis...@gmail.com on 12 Oct 2010 at 4:57

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Hi Regis, first I'd like to thank you for your excellent work. I really love 
this program, it's simple, integrated with Android build in dialer and can 
handle more than one account.

I got similar issue mentioned above, no sound while making calls. but it 
happened only with freephoneline.ca account. My another acount with voip.ms, 
worked fine. 

I'll try to get you some logs from both accounts. Hope it can help a little bit.

Original comment by den...@gmail.com on 18 Oct 2010 at 8:33

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@denghi : did you try to activate STUN? Most of the time audio in only one way 
is due to the lack of STUN and a NATed network (that was the cause of the 
initial issue).

However if you get absolutely no voice both way, that's probably something else 
than just a network traversal issue : 
Could you try when you reproduce the issue to toggle hold off and on again and 
say me if it helps? 
Some users has the same kind of issue (with desire) and reported me that 
holding and taking back makes things better.
If so I could probably try something (reverting a hack I did for Samsung Galaxy 
S support but done for all device since in theory it has no impact... but worth 
to try to do it *only* for galaxy S). 

Original comment by r3gis...@gmail.com on 18 Oct 2010 at 9:08

• Changed state: Fixed

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Amazing! after I set STUN server to 3cx's, everything is working fine now. I 
really appreciate your help. BTW, my device is HTC Desire from Canada.

Original comment by den...@gmail.com on 18 Oct 2010 at 9:33

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Great :)

I tried in 0.00-14 to activate by default STUN, but for many users it prevent 
from registering... so I reverted in 0.00-15 to default STUN disabled. 

However I'd like to find a way to inform users that if they are experimenting 
one way audio issue, it's probably due to the lack of a STUN server behind a 
NATed network.
I've already written something in the FAQ, but not easy to access for users 
when they are in the app... 
I'll probably do something within the app or add a webview in the app that 
display the wiki FAQ (or just a link).

Original comment by r3gis...@gmail.com on 18 Oct 2010 at 9:39

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Hi I'm having the almost the exact same problem as denghj. No voice on HTC 
Desire, in Canada, using freephoneline.ca as well. I tried activating Stun as 
well as ICE but still nothing.

Original comment by edwinho...@gmail.com on 20 Oct 2010 at 9:58

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@edwinhoang : could you try to hold/takeback the call to see if it's better 
when taking back. Somebody with desire reported me same kind of issue, but 
holding and taking back while in call solved the issue.

Anyway, if you can send me logs (see wiki : HowToCollectLogs ), could help me.
Thx

Original comment by r3gis...@gmail.com on 20 Oct 2010 at 10:11

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If "hold/takeback" is pausing and unpausing the call, yes I've tried it. I've 
also tried turning ICE on. So far nothing. I have sent you a copy of my log. I 
hope you can find a solution.

Original comment by edwinho...@gmail.com on 20 Oct 2010 at 10:40

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@denghj and also @enciso & @edwin => I've opened an issue specific for 
Freephoneline.ca (I guess you three are using this provider) : issue 296.

If you succeed in making it working could you share the configuration you used 
on the issue 296? I'll try to add a wizard for this provider that automatically 
fill fields. But as far as I understood denghj you succeeded in making it 
working activating stun, however edwin doesn't succeed even if he activates 
STUN. If enciso could try with stun and give us feedback too could be nice ;) 
Thanks in advance all !

Original comment by r3gis...@gmail.com on 22 Oct 2010 at 11:25