#669 no incoming sound

[Anonymous]

Originally created by: yosef.d...@gmail.com

What steps will reproduce the problem?
1.no incoming sound
2.
3.

What is the expected output? What do you see instead?

What version of the product are you using? On what operating system?

2.2 android
Please provide any additional information below.

I use galaxy i9000 on wifi works well on 3 g   ican call other party hear me
i cant hear the other party tried stun doesnt help to resolve
the problem

[Anonymous]

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

Sorry I didn't see your mail. I'll investigate thanks for logs ;)

Status: New

[Anonymous]

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

I & others using my ISP have been experiencing this issue for quite a while. It seems to be quite a random thing.
From my point, i am using huawei u8150 phone with Android 2.2 froyo.
Here is a link to a little information posted on my isp user forum
https://forum.exetel.com.au/viewtopic.php?f=308&t=37609

[Anonymous]

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

What could be interesting would be to reduce number of codecs activated in CSipSimple configuration which will reduce packets size. (long press on a codec row to deactivate)

I don't think that's the root of this problem but could be interesting to test.

[Anonymous]

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

I actually already have reduced to only use 2 codecs as pcma & pcmu a long time back. ( This is the case for both narrow & wideband set up )
Also forgot to mention before, my connections for fixed adsl & WBB both use static IP addresses from my provider. The adsl is nat'd but the wirelessBB is not. Using STUN, ICE or both makes no difference. my provider also supplies the SIP service we are using and have no port blocking for voIp as they are actively encouraging us to use their voip service & in fact have written their own app for use on iphones.
Also while i think of it, your CSipsimple application is by far the best mobile Sip client I have tried to date. If this little nag could be fixed it would wonderful.

[Anonymous]

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

Hi, Managed to catch this error into a log which I have now forwarded to your 'developers address.
Hope this helps.

[Anonymous]

Originally posted by: yosef.d...@gmail.com

i tried to canel the other codecs and it didnt work still dont have incoming
sound,on siproid it works fine withiut problem.but i prefer csipsimple
please help me to resolve this issue

[Anonymous]

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

Any resolution to this?  I see the exact same problem with Sipgate, showing no incoming audio.  Strangely, the Sipgate echo test line works fine.

[Anonymous]

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

I have the same problem on Sipgate.de

What happens is as follows:

ON AN INCOMING CALL:
- no incoming audio from the caller
- outgoing audio from my side works (ie. the caller can hear me).

ON AN OUTGOING CALL:
- both sides audio works.

This is on a HTC Sensation using the gingerbread daily build (7/21).

[Anonymous]

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

I have this same problem even when using WiFi, and the problem did not occur when I first installed the application (i.e. everything used to work fine). 

I have updated, but unfortunately I do not recall what the old (working) version was or precisely when the problem started.  To reiterate, I have acceptable audio on outgoing calls, but for incoming calls I cannot hear anything, however apparently the incoming caller can hear me. 

This is on a Nexus one running CM7.

[Anonymous]

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

According to the FAQ, the problem with lost incoming audio can be fixed with a STUN server.  Has anyone tried setting up STUN and seeing if it resolves this issue?

[Anonymous]

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

@guru : if incoming and outgoing does not behaves the same way it could also be linked to the codec negociation.
So servers does not support very well codec renegociation and it may lead to receive audio in a codec not expected by the application.
To check if this is the problem, a good test is to disable all codecs but PCMU and see how it goes.
Then, you should select the best codec supported by your sip provider as the only available codec in csipsimple.

[Anonymous]

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

I'm having the same issue on 3g, though all is perfect on wifi.

Tried different codecs... Only 8kHz codecs work.

Sipdroid does work really well using PCMA/U 64kbit on same 3g.

Tried STUN, ICE. And a couple different stun servers.

These are for incoming calls. Gv/iptel on 2.3.4gb 45.2.17 bell.

Any other things to try?  Thanks!

[Anonymous]

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

Same problem here, only on incoming calls on 3g.

Also tried the guide online to use STUN and ICE, doesn't help.

[Anonymous]

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

Issue 673 has been merged into this issue.

[Anonymous]

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

This problem is usually due to codec negociation on server side.
A good test is to disable all codecs but PCMU for example.
If one that has this problem can try with a nightly build
http://nightlies.csipsimple.com/trunk/
with only STUN enabled and only PCMU codec enabled and collect some logs with this issue number as reference (see HowToCollectLogs wiki page).

Status: Need-Details

[Anonymous]

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

Issue 581 has been merged into this issue.

[Anonymous]

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

Great news. Thanks for the feedback :)

Status: NextRelease

[Anonymous]

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

Sorry r3gis...@gmail.com I spoke to soon.

Strangely enough the first time I tested it seemed to have been fixed, but since, I have not been able to make it work at all.

I don't think this is an codec negociatio problem. I think it relates instead to the incorrect routing of audio set by the ROM on my phone. (ZTE Crescent) Theres more details on a similar problem with various handsets including the ZTE blade over @ sipdroid where with their latest release, the audio is being sent out of the Rear loudspeaker rather than earpiece.

https://code.google.com/p/sipdroid/issues/detail?can=2&start=0&num=100&q=&colspec=ID%20Type%20Status%20Priority%20Milestone%20Owner%20Summary%20Stars&groupby=&sort=&id=366

Although it seems the problem is related to a broken rom, It would be nice to see a way to set the output speaker mode, similar to the mic input mode as you can in advance preferences.

[Anonymous]

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

According to the working patch in sipdroid issue list the option to turn on in CSipSimple is "use mode API".
I think there is no need of extra tweaking options than the ones we already have.
Somebody on the sipdroid mentioned a working work around in csipsimple. I'll search in history what were working settings to share.
Normally when there is working workaround I add auto detection for the device so that any body benefit of that.
Can you send me logs from your device (see HowToCollectLogs wiki page) so that I can see the device model.
Also after switching to use mode API,try to reboot the phone.If it's a rom bug they don't take into account reset made by apps.

[Anonymous]

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

Known values for ZTE blade :
Use Mode API = true
Audio mode while in call = MODE_IN_CALL

Apparently some other values for ZTE Joe are necessary :
https://code.google.com/p/csipsimple/issues/detail?id=1736#c4
Use Mode API = true
Use Routing API = true

[Anonymous]

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

r3gis...@gmail.com thanks for your ongoing help. It is fantastic.

This is a seperate issue from Sipdroid.

With Sipdroid, Audio is routed inccorectly through rear speaker, with CSip, no audio is heard at all.

This is what I tried after what you said. Convinced I had tried all these options before, including the Samsung Galaxy S hack, i thought I best give it a go again, but this time log my doings:

1-
Use Mode API = true
Use Routing API = true
Audio mode while in call = MODE_IN_CALL
Reboot
Test Call
Result= No Incoming Audio through Earpiece

2-
Use Mode API = true
Use Routing API = FALSE
Audio mode while in call = MODE_IN_CALL
Reboot
Test Call
Result= No Incoming Audio through Earpiece

3-
Use Mode API = true
Use Routing API = true
Samsung Galaxy Hack = true
Audio mode while in call = MODE_IN_CALL
Reboot
Test call
Result= No Incoming Audio through Earpiece

4-
Mode API = true
Use Routing API = true
Audio mode while in call = MODE_IN_COMMUNICATION (Default)
Reboot
Test Call
Result= No Incoming Audio through Earpiece

5- (Default Set Up)
Use Mode API = FALSE
Use Routing API = FALSE
Audio mode while in call = MODE_IN_COMMUNICATION (Default)
Reboot
Test Call
Result= No Incoming Audio through Earpiece

I tried mode in call = Normal and RINGTONE too. No change.

I sent the log, but here it is also.

Based on GPL application CSipSimple version : 0.04-00 [r1513]
Here are important informations about Device :
android.os.Build.BOARD : blade2
android.os.Build.BRAND : ZTE
android.os.Build.DEVICE : blade2
android.os.Build.ID : GRJ22
android.os.Build.MODEL : Blade S
android.os.Build.PRODUCT : P736V_FFR
android.os.Build.TAGS : release-keys
android.os.Build.CPU_ABI : armeabi
android.os.Build.VERSION.INCREMENTAL : eng.zhangchun.20111208.093059
android.os.Build.VERSION.RELEASE : 2.3.5
android.os.Build.VERSION.SDK_INT : 10

----

No Am I tested this right? Because when I perform the Audio test, the earpiece works FINE!!

[Anonymous]

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

Also want to add the model is listed as the Blade2 / Blade S

That is the french market name for the ZTE Crescent from my ROM. It is indeed the ZTE Crescent that I am using.

[Anonymous]

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

MMmhh, it's maybe something with network NAT traversal or with codec.
Can you :
* ensure that stun is activated (only stun not ice)
* try to disable all codecs but PCMA/PCMU for both fast and slow networks

[Anonymous]

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

Unfortunately that doesn't help.
Sipdroid gives me no problems with the earpiece, despite no stun server set. So surely it can't be a related to codecs negotiation/network problems?

[Anonymous]

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

On another note I get massive delay using PCMA both in Csip and Sipdroid. I never had this before but it started happening all of a sudden. Its not a problem because I GSM gives me no delay.

I just want to find out why Csip I don't get incoming audio! Annoying!