DTMF problem starting with 1.3.8

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1. Install or upgrade to 1.3.14
2. Call or respond
3. On a Nexus One with sipdroid I hear the other side perfectly, but the
other side hears nothing.

What is the expected output? What do you see instead?
To be able to hear eachother.

What version of the product are you using? On what operating system?
1.3.14

Which SIP server are you using? What happens with PBXes?
Asterisk

Which type of network are you using?
WLAN

Please provide any additional information below.

Original issue reported on code.google.com by disco...@gmail.com on 17 Feb 2010 at 7:11

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To analyze the problem it would be helpful to know
1) if this happens with PBXes as well?
2) does the other side really hear nothing, or just parts of speech?

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 8:14

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1) I've deleted my PBXes configuration, so I'm not able to test that presently.
2) Further testing reveals that the audio is severely garbled but is getting 
through

Original comment by disco...@gmail.com on 17 Feb 2010 at 8:30

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Please try attached image.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 8:35

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I just made another test, this time the audio was clear, but _extremely_ low 
volume.
  Were any changes made which could effect the volume level?

Original comment by disco...@gmail.com on 17 Feb 2010 at 8:37

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Volume gain can be adjusted in advanced options. What about the test image, can 
you 
reproduce any change regarding the garbled audio?

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 8:43

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Or is garbled audio maybe related to screen on/off?

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 8:44

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The volume gain was fine before.  I don't have a landline so I have to use my 
voicemail 
system to record messages to test.  Its a bit painstaking as now I have to try 
repeatedly to get it to understand the dtmf (this worked fine before).   I 
can't seem 
to get that apk to install.   I have unknown sources enabled, and I downloaded 
it 
directly to the device but it said download unsuccessful.   

From the pc, adb install seems to just hang forever...

Original comment by disco...@gmail.com on 17 Feb 2010 at 8:57

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Normally you would not need to uninstall first, bit in this case it is worth a 
try.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 9:04

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Voicemail is not a good test (except if it is an old tape recorder!).

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 9:06

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Another image to try (please don't mix them, they have two different changes).

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 9:16

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I am having a very similar issue only with incoming audio on outbound calls. 
Same
version of sipdroid Gizmo5

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 9:17

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I live in Japan and my VOIP line is for my US number.   Way too early to call 
anyone 
in the US.  Calling my cell phone from my cell phone is useless and my wife is 
at 
work.   I usually leave the voice messages or also test with 1800GOFEDEX ;)   

I got that new apk installed (downloaded it to my own http), I think the phone 
just 
didn't like pulling it out of the forum url.  No idea why adb install didn't 
work.

No change, though.   I just tried increasing the microphone gain to highest, no 
difference.   The fact that the DTMF tones aren't registering consistently 
suggests 
its a deeper issue than mic gain.   

Is there a way I can access the older versions so I can confirm that it was 
working 
before?   There have been a couple of new releases recently, I'd like to make 
sure I 
can pinpoint exactly when it started, and be double sure there isn't anything 
"weird" 
happening with my handset or Asterisk server.

Thanks!

Original comment by disco...@gmail.com on 17 Feb 2010 at 9:20

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Which version have you been using before the upgrade?

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 9:34

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I always use the latest.  Its possible I didn't test audio with 1.3.13, though. 
Maybe if you could put up at least 1.3.12 and 1.3.13 if it isn't too much 
trouble. 
Once this is resolved I'll be sure to always fully test each release and keep a 
copy
since they're not retained on the google code site.

Thank you.

Original comment by disco...@gmail.com on 17 Feb 2010 at 9:58

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I think the critical change was ealier, but let's try.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 10:02

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Sigh you may be right.  Recently I was waiting for a different issue (where 
when the
call exists the next time you run dialer it dies) to be resolved and I guess I 
didn't
actually punch in my VM password.   Which version do you think the relevant 
change
would've been made?   Going by my call log, my best guess for a certain 
confirmation
of it working was Feb 5. =(

Original comment by disco...@gmail.com on 17 Feb 2010 at 10:11

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OK, two more.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 10:36

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If I switch to tcp outbound calls work better... The party called can here me 
ok but
their audio is garbled ... As apposed to not working at all. 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 10:39

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All calls have one way audio using the stun server stun01.sipphone.com ...  udp 
is
better than tcp on my "audio incoming" on "outbound calls".. 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 10:46

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Thanks for posting those.   I feel like I'm going crazy, I could have SWORN that
1.3.10 worked fine.  I switched from Asterisk 1.4 to 1.6 so that I could use 
TCP on
Feb 3.  I would have tested it at that point.   But....

1.3.7 -works-
1.3.10 and later do not work

Original comment by disco...@gmail.com on 17 Feb 2010 at 10:46

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Have you tested above trial versions already? Also included 1.3.8 and 1.3.9 to 
finally 
pinpoint this. Thanks for testing this asap.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 10:54

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All tested.   1.3.7 works.   1.3.8 and later do not.

Original comment by disco...@gmail.com on 17 Feb 2010 at 11:04

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Version 1.3.8 did some changes to DTMF. Can you test this without dialing a 
DTMF digit 
before speaking?

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 11:17

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You're right.   Voice works up to and including 1.3.12.  Probably explains my
confusion, there are two issues. 

1.3.8 DTMF stops working consistently
1.3.13 Outbound voice is degraded

Original comment by disco...@gmail.com on 17 Feb 2010 at 11:28

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1.3.13 had a problem in outbound voice. Check 1.3.14.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 11:32

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Ok, first off I didn't notice before you posted two unnumbered Sipdroid.apk 
images. 
I did some more testing of 1.3.14 and also the second Sipdroid.apk.   It seems 
that
1.3.14 improves upon 1.3.13 but still has voice issues.   The Siproid.apk is 
better,
but also it seems there is something going on after 1.3.12 that is still 
present.  
My wife will be home in a bit and I'll try calling her from my SIP phone so 
that I
can test more completely.   My test is admittedly crude, although if I call 
FedEX
with 1.3.7 through 1.3.12 and say "international services", the computer can
understand me 100% of the time.   1.3.13 it can't understand me, 1.3.14 its 
about
40-50%, and the seccond Sipdroid.apk I get about 60%.  Hopefully with the second
phone I'll be able to better describe what I'm hearing.

Original comment by disco...@gmail.com on 17 Feb 2010 at 11:46

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I have the same problem on the Motorola Milestone with the outbound audio!

Original comment by Alex.Herrmann@gmail.com on 17 Feb 2010 at 11:50

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I have garbled audio on outbound calls on the outbound audio settings "both
tcp/udp"... I have NO audio on the outbound audio of an outbound call when stun
server is enabled "both tcp/udp". 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 12:29

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1.3.7 Also fixes issue for me. 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 12:45

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Brandon: what about 1.3.12?   It turns out I was having a separate DTMF related 
issue
with 1.3.8, but 1.3.12 seems okay audio wise

Original comment by disco...@gmail.com on 17 Feb 2010 at 12:48

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On my father's N1 I've verified there is no more audio issue on 1.3.14. Version 
1.3.8 
changed DTMF from SIP INFO to RFC 2833 discoltk is having a problem with.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 12:53

• Changed title: DTMF problem starting with 1.3.8

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Forgive me 1.3.12 is not in the download tab. I am no longer able to see the
attachments. Can some point me to the correct location... I would also like to 
point
out the Stun server comments I made earlier. I believe this to be related. 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 12:55

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From sip show settings:

  DTMF:                   rfc2833

I'm not sure about the sound quality issues, trying to call my wife's phone but 
I'm
getting congestion from my SIP provider on the international call.  I normally 
use it
for US bound traffic.  I'll try to call someone before I go to bed once people 
are
awake in the US.

Original comment by disco...@gmail.com on 17 Feb 2010 at 1:08

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What was the nature of the change between the second Sipdroid.apk you posted 
here and
the 1.3.14 release?   I switched back to 1.3.14 and I can't get FedEX computer 
to
understand me much at all, but it works better with the second Sipdroid.apk 
(better
yet with 1.3.12).

Original comment by disco...@gmail.com on 17 Feb 2010 at 1:20

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I would like to get a copy of the second sipdroid.apk that is working to test. 
Any
idea how where I can download a copy of this? 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 2:33

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I changed a buffer size for the second. It can be downloaded again.

Original comment by pmerl...@googlemail.com on 17 Feb 2010 at 2:50

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@brandonnolte:  It is attached at comment #10

I need to do further testing to try to quantify the difference in audio between
1.3.12, 1.3.14, and the "Siproid.apk" from comment 10.  

Original comment by disco...@gmail.com on 17 Feb 2010 at 2:58

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I used the sipdroid.apk in comment 10. Still sounds terrible. 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 3:46

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How about 1.3.12?

Original comment by disco...@gmail.com on 17 Feb 2010 at 3:49

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1.3.12 works fine... 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 3:58

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Is there anyway we can get an adjustable jidder buffer. 

Original comment by brandonn...@gmail.com on 17 Feb 2010 at 4:00

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@brandonnolte:  You're using Nexus One, correct?

Original comment by disco...@gmail.com on 17 Feb 2010 at 4:04

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Yes. I am on the n1 phone with a gizmo5/GV setup. 

Original comment by brandonn...@gmail.com on 18 Feb 2010 at 6:35

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I was able to do some more testing with a live person, it seems the sound was 
okay
with the Sipdroid.apk from comment 10, but I think 1.3.12 is still the last 
time it
was 'normal'.

Original comment by disco...@gmail.com on 18 Feb 2010 at 6:41

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This issue sounds similar to 
http://code.google.com/p/sipdroid/issues/detail?id=326 I
am curious if the fix issued in 
http://code.google.com/p/sipdroid/source/detail?r=467
fixes our problem. I would love to test it out. 

Original comment by brandonn...@gmail.com on 18 Feb 2010 at 7:49

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Have you tested with a stun server? 

Original comment by brandonn...@gmail.com on 18 Feb 2010 at 7:59

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@brandonnolte:  Actually I lazily copied the exact text from that case.  When I
updated to 1.3.14 I read what it fixed, and then tested to make sure everything 
was
okay.  That was when I discovered the quality issues, which seemed very similar 
to
issue 326.  I guess I hadn't actually tested 1.3.13.  I'm fairly certain there 
is
still something wrong in 1.3.14 on Nexus One.

STUN is for UDP only.  Before Sipdroid implemented STUN I was having trouble 
with
NAT, so I upgraded Asterisk to 1.6, which supports TCP.  Now that I use TCP I 
don't
have the NAT troubles, and don't need STUN.  So, no I haven't tested it.

It seems that we need more people to confirm our experience with 1.3.14 sound on
nexus one, as the developer changed the title of this issue to only reflect the 
DTMF
problem.  He tested on an N1 and says he had no problems. 

Original comment by disco...@gmail.com on 18 Feb 2010 at 8:29

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On 1.3.14, I get incoming calls only on WiFi, not 3G.  Lot of static when not 
using
stun server.  Static goes away if I use a stun server.  On an incoming call, I 
have
outgoing audio working, but don't have incoming audio.  I am using a Nexus One. 
Sipdroid is set up to use Sipgate and Sipsorcery along with Google Voice.

Original comment by rpras...@gmail.com on 19 Feb 2010 at 6:51