Sound problem using asterisk / standard sip phones

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What steps will reproduce the problem?
1. Install imsdroid-2.0.395-preview.apk (Samsung Galaxy tablet )
2. configure to register into an asterisk server
3. call an outside line or another extension in the asterisk server (using 
polycom phones )

What is the expected output? What do you see instead?
normal conversation / sound coming out of the speaker (IMSDroid) is normal, 
sound being transmitted to the outside call or the sip extension is very choppy 
( not acceptable at all ).

More info.  I installed the boghe soft phone into a win7 computer / registered 
into the same asterisk server and called the extension the boghe softphone is 
configured.  All sound is perfect / even video is great.  The boghe phone 
sounds perfect calling outside phones or other sip extensions on the asterisk 
server.  

So.  boghe works perfect with asterisk, IMSDroid seems to have a problem 
talking to outside calls or other sip phones on the same system. I do not have 
these problems using SipDroid.

I posted this problem some time ago and other people have verified it. I was 
hoping ver 2 fixed the problem but apparently not.

Any advise?
Thanks

Original issue reported on code.google.com by jdpflor...@gmail.com on 21 Apr 2011 at 4:43

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could you please attach network trace (both sip and rtp)?

Original comment by boss...@yahoo.fr on 21 Apr 2011 at 7:16

• Changed state: Accepted

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can you suggest a way  to get the trace?  my galaxy tab is not rooted and I do 
not wish to  void the warranty.

Original comment by jdpflor...@gmail.com on 23 Apr 2011 at 2:14

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[deleted comment]

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I have something to add here.  I changed IMSDroid settings to use g729 ( exact 
same asterisk server ) and the voice quality is now great.  Switch back to PCMU 
and the horrible voice quality is back.  So I believe it could be a codec 
problem.

One more question, where did DTMF go?  While on a call you no longer can get a 
keyboard to send DTMF.

Thanks

Original comment by jdpflor...@gmail.com on 23 Apr 2011 at 7:03

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Very strange because G.711 is used by 90% of the users and I've never received 
such feedbacks. Does the RTP packets delivered p2p or through your asterisk 
server?
You can make capture from the first node after your mobile (e.g. proxy server).
You are right DTMF is missing in this preview version but will be re-added in 
the next ones.

Original comment by boss...@yahoo.fr on 28 Apr 2011 at 12:52

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Hi  , 

Have the same problem with G711 codec and sip phone whith low rtp audio buffer 
( jitter compensation ) . On the network dump , we can see a few jump on G711 
stream. Have resolved this by fix timing of the audio pump . ( source-MIC- -> 
rtp stack ) .  I'm not sure that is the good solution , but this patch fix 
audio quality for me. If you can confirm ?
patch apply to IMSDroid branche V2 .
Best regards
bye , 
Philippe  

Original comment by verney.p...@gmail.com on 12 May 2011 at 5:28

Attachments:

• imsdroid_fix_audioProducerTiming.patch

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Original comment by boss...@yahoo.fr on 5 Jun 2011 at 5:30

• Changed state: CannotReproduce

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I also meet this same prob. I only use the g729 codec,Sip Call party hear the 
voice is choppy，and I add the g711,the call party hear the voice is very 
clear,but the called party is very noise

Original comment by santongs...@gmail.com on 31 Aug 2011 at 7:27

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I also experienced noise problem on the other side of the call when I used my 
LG Optimus Black P970. The problem are on both incoming & outgoing calls, and 
on 711/729 codec.
However, the problem does not happen on Motolora MB525.
Both P970 and MB525 runns 2.2.

Original comment by mingwaic...@gmail.com on 27 Sep 2011 at 4:08