Can't hear other person

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What steps will reproduce the problem?
1. dial other person
2. listen

What is the expected output? What do you see instead?
The other person can hear me and the audio quality seems to be very good
(woohoo!). I can't hear the other person talking at all

What version of the product are you using? On what operating system?
Sipdroid 0.9 + the official ADP 1.5 Image with current radio fw

Please provide any additional information below.
Tried using sipgate.de directly and pbxes.com (with sipgate)

Original issue reported on code.google.com by marc.see...@gmail.com on 28 Apr 2009 at 8:53

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Did you try from different networks? Maybe this is related to your router.

Original comment by pmerl...@googlemail.com on 28 Apr 2009 at 11:38

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Ah, it could be due to the missing port forwarding.
Using 3G, it has a huge lag, but 2 way communication works

Original comment by marc.see...@gmail.com on 28 Apr 2009 at 12:20

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I have same issue, audio quality appears to be good for outbound from phone, 
but I am
unable to hear any incoming audio.

This is running on a private wireless network, I have no sim installed in the 
phone.

FW V1.5

I have tried using connection via PBXes.org which solves problems with 
authentication
on different phones, but whether I use PBXes or login direct to my sip provider
(AQL.com) the same issue occurs, they hear me fine but i hear nothing.

Original comment by mark.boo...@gmail.com on 29 Apr 2009 at 6:10

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Another reason might be volume set too low. Try pressing the volume buttons 
while in 
call.

Original comment by pmerl...@googlemail.com on 30 Apr 2009 at 12:38

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I checked the media volume and this is set to max.

I also noticed that the call is dropped by the G1 after about 10 seconds, I am
guessing this is because it does not see any inbound audio?

Any other thoughts?

2 way dialing works fine.

Original comment by mark.boo...@gmail.com on 30 Apr 2009 at 10:45

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i have the same issue, with 1.5 and sipdorid the latest verison.

i cant hear the other side.
the other side cant hear me.
Any solutions?

Original comment by mni...@gmail.com on 1 Jul 2009 at 5:32

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I can connect via WLAN or 3G to voipcheap.com. I can ring a number, and it 
connects
the call, however no sound either way. 100% sure that the call is established, 
just
no audio what soever. Shame, cos this would be ideal for me in the house. Hope 
you
guys find the issue. BTW I have an HTC Magic.

Original comment by miles.ba...@gmail.com on 2 Jul 2009 at 2:37

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I have an HTC Magic and tried every thing possible, it seems that there is a 
hardware barrier which does not allow 
either sides to hear but call connection is perfect.
Sipdroid needs to be improved alot, i actually wasted my money on buying the 
HTC Magic, all apps are free but 
mostly dont work well.
Already sold it and bought a new Iphone 3gs, everything works out of the box

Original comment by mni...@gmail.com on 12 Jul 2009 at 6:37

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After upgrading to 1.0.1, can no longer hear calls, though I can be heard. 

Connecting through GV to gizmo5 proxy01.sipphone.com. 

Outgoing calls work fine.

Is this a problem with Gizmo5 / GV / Sipdroid?

Original comment by imaginea...@gmail.com on 15 Jul 2009 at 9:14

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I'm also having the same issue. I can answer calls but we can't hear each other
talking. Obviously this is a roadblock to using this VoIP solution, hopefully 
it will
be fixed soon. If any more information on my setup would help solve the issue 
please
let me know.

Original comment by joshua.p...@gmail.com on 18 Jul 2009 at 3:30

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I'm on sipdroid 1.0.2 I can establish a call and hear the other person, but the 
other
person can't hear me.

Original comment by joachim....@gmail.com on 22 Jul 2009 at 9:21

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Issue 90 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 24 Jul 2009 at 2:40

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To add more information about my setup (see comment 10), I am using my 3G 
network
connection exclusively when trying to have a sipdroid call. I am running 
Android 1.5,
my gizmo phone number, and Google Voice is forwarding the original call to my 
gizmo
number that is being answered by sipdroid. I can have a successful call when 
using
Ekiga on my laptop (in place of sipdroid on my cell phone).

Original comment by joshua.p...@gmail.com on 24 Jul 2009 at 3:21

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i have post a same problem.... Issue 82

Please Manage to solve it fast..

Original comment by trushs...@gmail.com on 28 Jul 2009 at 10:18

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same problem

Original comment by geoff.si...@gmail.com on 5 Aug 2009 at 10:15

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I had similar problem. Now I've solved it.

The other side couln't hear me but I could hear them.

Sipdroid uses only G711 codec.. so

I modified some code.

in /SipUA/src/org/sipdroid/media/JAudioLauncher.java
42   int frame_size=125; // modified 500 to 125
44   int frame_rate=64; // modified 16 to 64

in /SipUA/src/org/sipdroid/media/RtpStreamSender.java
116  this.frame_size = frame_size; // modified 1024 to frame_size

according to
http://www.comsoc.org/livepubs/surveys/public/2004/apr/figures/scheets-table-1.h
tml

cupcake 1.5
sipdroid 1.0.5
imtel.com sip service provider

Original comment by overjoo...@gmail.com on 25 Aug 2009 at 12:10

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hi overjoowon

i have an issue where i can established a call and talk with 1 provider, 
however using 
another provider the remote phone rings but there is no sound both ways. i 
checked the 
codecs used by this provider and they say they support G711A & 711U. i'm 
wondering if 
you had a similar issue or could suggest a fix for my problem.

Original comment by galvat...@gmail.com on 27 Aug 2009 at 6:52

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I have same problem on HTC Hero (root).
Provider : freephonie.net
Connection type : WiFi (good signal)

Incoming calls make ring my phone but I cant hear anything, It's the same when I
call. I listened only one time my correspondent on many tests.

Thanks you,
Benjamin

Original comment by benjamin...@gmail.com on 28 Aug 2009 at 7:16

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I have this exact same issue too.
Here's the details I have:

Model: HTC Dream 
Firmware: V1.5
Baseband version: 62.50S.20.17H_2.22.19.26I
Kernel version: 2.6.27.24-cm (shade@toxygene)
Build Number: htc_dream-eng 1.5 CUPCAKE eng.benji.20090609.153832 test-keys
Root: The Dude

Using: SIPDROID v1.06
Server: PROXY01.SIPPHONE.COM
Port: 5060
Protocol: UDP
Using: WLAN (Full Signal Strength)

I use my Gizmo5 account in conjunction with Google Voice to make calls over 
WIFI.

For incoming calls my Google Voice is set up to forward all calls to Gizmo5.

For outbound calls my Google Voice calls my Gizmo5 to put it on the line, then 
calls 
the recipient.

When connected: the recipient says they can hear me but all I hear is silence.  
After 
about 10 seconds of silence the line just disconnects.

If I go to the web based Phone or Gizmo Software on my computer, I can answer 
the 
call and talk.  The problem seems to lie either with the phone or SIPDROID.

This particular issue began a few days ago when I updated a bunch of my apps.  
Before 
that, I was been able to make calls over WiFi.  Two possible apps that could 
have 
caused this problem is the updated SIPDROID or Google Voice.

I hope this problem gets solved, and that I contributed in any way.

Respectfully,
Jordan

Original comment by foley...@gmail.com on 2 Sep 2009 at 2:44

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I was having a similar problem with an ATA (_not_ sipdroid), and ended up 
having to
configure STUN. Outbound worked correctly without STUN, but I couldn't hear any 
audio
from inbound callers (to my Gizmo5 account via the ATA) until I configured my 
ATA to
use STUN. 

Original comment by kevin.lo...@gmail.com on 2 Sep 2009 at 11:46

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overjoowon's solution worked for me, if others want to test there is a patched 
apk at 
http://caxica.freeshell.org/android/Sipdroid-debug.apk 

Original comment by serge.de...@gmail.com on 3 Sep 2009 at 1:39

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is Sipdroid-debug suported (or just teasted) by dev team ? 

Original comment by benjamin...@gmail.com on 3 Sep 2009 at 7:46

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it is the standard sipdroid 1.0.5 with the modifications mentioned by 
overjoowon. I 
will build an updated one with 1.0.6 which was just released. I put it out to 
see if 
other people had success with it using other providers. I use Betamax and the 
standard 
version does not work for me. If this one works for you then we should try to 
get the 
developers to accept the patch or fork the package if they do not want to.

Original comment by serge.de...@gmail.com on 3 Sep 2009 at 9:31

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I just tried it them and it works perfectly, I am going to stay using these 
development versions as they are much better than the restricted standard 
versions.

Original comment by wom...@gmail.com on 3 Sep 2009 at 10:00

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I can confirm that the overjoowon's solution works for me as well! It has fixed 
the
problem what I have described here (ad "2)"):
http://groups.google.com/group/sipdroid-developers/browse_thread/thread/24d4aa57
e65725f1/657a2396d5049c40?lnk=gst&q=cesnet#41b86625f3bb430d

My SIP provider is CESNET (http://sip.cesnet.cz).

Original comment by jiri....@gmail.com on 3 Sep 2009 at 10:26

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I have added the proposed patch into release 1.0.7. Sipdroid now sends 160 
samples 
which is the commonly used frame size (see other issue).

Original comment by pmerl...@googlemail.com on 3 Sep 2009 at 8:29

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Issue 82 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 5 Sep 2009 at 8:40

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Issue 93 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 5 Sep 2009 at 8:41

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[deleted comment]

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Issue 123 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 6 Sep 2009 at 11:33

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Sipdroid 1.05 to 1.07 works fine with asterisk 1.4.17 beeing user 7022 as long 
as I'm
in the same subnet than the server (192.168.0.0). But when I login from remote 
then
the data packets channel goes to localhost instead of the originating global 
address.
See sip channels below. Usually this problem is solved with a stun server.  

82.136.75.59     7022        3e507bf33f9  00102/00000  0x0 (nothing)    No      
Init: INVITE
192.168.2.30     702         1f8931ab405  00102/00000  0x4 (ulaw)       No      
 Tx: ACK
127.0.0.1        7022        34496635085  00101/00002  0x8 (alaw)       No      
 Rx: ACK

Hope this helps

Etienne

Original comment by etienne....@gmail.com on 7 Sep 2009 at 8:02

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Any progress on this? It doesn't bother me too much (just want to see when I 
get a
call where I have wifi but no data), but it would be nice if it worked.

Gizmo5 and GV here as well. G1, Cyanogen mod.

Original comment by dlagesse...@gmail.com on 8 Sep 2009 at 12:47

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Issue 51 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 10 Sep 2009 at 8:10

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The data packets going to the localhost is probably due to the same reason that 
as
Issue #9, the program reporting its public IP as 127.0.0.1.

Original comment by uborst...@gmail.com on 17 Sep 2009 at 6:21

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Same here ... unable to use sipdroid cos i cannot hear anything ...

Original comment by lprassa...@gmail.com on 2 Nov 2009 at 5:14

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Issue 174 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 6 Nov 2009 at 10:52

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Issue 195 has been merged into this issue.

Original comment by pmerl...@googlemail.com on 14 Nov 2009 at 12:46

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I am having a similar issue.
HTC Hero 1.5, running MoDaCo 3.9
SIPDroid 1.1.8
SIP Provider: Globe7
Proxy: 84.45.70.14/15

The call gets established but on answering the call, there is no audio either 
ways. I
cannot hear the other party nor can the other party hear me. I cannot even hear 
the
ring while calling.

Please HELP!

Original comment by bajaja...@gmail.com on 15 Nov 2009 at 8:44

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I had similar problem. Now I've solved it.

The other side couln't hear me but I could hear them.

Sipdroid uses only G711 codec.. so

I modified some code.

in /SipUA/src/org/sipdroid/media/JAudioLauncher.java
42   int frame_size=125; // modified 500 to 125
44   int frame_rate=64; // modified 16 to 64

in /SipUA/src/org/sipdroid/media/RtpStreamSender.java
116  this.frame_size = frame_size; // modified 1024 to frame_size
==================================================================

How can I change this code ? I need a software ? Tkanks for your response on
olivier.verateam2@gmail.com

Original comment by olivier....@gmail.com on 21 Nov 2009 at 10:29

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I'm seeing the same problem with sipgate.com. I'm unable to effectively make 
outgoing 
calls with sipdroid, due to one-way voice (sipdroid receives "scratchy" voice 
from 
called party, but does not transmit).  Incoming calls work perfectly.   Placing 
an 
outgoing call using sipgate's web site works as well, though this is 
effectively an 
incoming call as far as sipdroid is concerned.

Everything works fine with pbxes (albeit with some reliability issues), but I'd 
prefer not to add another party to the equation.

@oliver.vereteam2: You'll need the source from SVN and the Android SDK.  Then 
follow 
the directions in the BUILD.txt file.  You'll also need Java and Apache Ant 
installed 
on the machine.

Original comment by tlieb...@gmail.com on 22 Nov 2009 at 1:56

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I'm having this problem also, clearly appears to not support NAT.

Nexus One running factory 2.1 firmware
Siproid 1.3.5beta
SIP Provider:  Run my own asterisk server

This works fine on my siphone app on iphone 3g, it seems pjsip deals with 
several
solutions for NAT. including but not limited to STUN:

http://www.pjsip.org/pjsua.htm

Original comment by disco...@gmail.com on 21 Jan 2010 at 11:46

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RE comment 44: FWIW, I have successfully used Sipdroid with both a Droid and 
Nexus
One with NAT. On the Droid, I have very few issues. The Nexus One had a lot of 
issues
with call quality, but not with NAT, in my experience.

Original comment by davidshi...@gmail.com on 22 Jan 2010 at 12:01

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Maybe I have the same problem on Nexus one original ROM, latest Sipdroid, 
registers on 
Wlan and 3g , I can call others , where we hear each other just well, But when 
I receive a call we cant't hear each other. the same problem is there in Fring 
using the 
same sip-provider -alltele.se-( tested the account using twinkle on the same 
LAN and it receives calls just fine) so something here is conflicting with 
Android 2.1 

Original comment by shwan.ciyako@gmail.com on 25 Jan 2010 at 8:19

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Re: comment 45:  Thanks for the feedback.   Are you using PBXes or your own SIP
server?   The sipdroid FAQ mentions NAT is supported on PBXes, but I'm not sure 
why
it wouldn't work with my asterisk server.   I'm able to use it with siphone, and
other softphone clients that are behind NAT.   There is no place to configure 
STUN,
so I thought maybe PBXes has STUN hard-coded internally the app?

Original comment by disco...@gmail.com on 3 Feb 2010 at 2:38

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Ok I did some more testing.  Sipdroid may have been updated and fixed some 
quality
issues with the N1 since I first tested, which may have been part of the 
problem.  In
the asterisk console, it is still showing my private address.  This is not the 
case
when using other clients.  Previously I had been able to make calls, but the 
other
end was unable to hear me.  I usually test by calling Fedex and speaking with 
their
computer.  Now it seems to work, although it is strange that "sip show 
channels" 
reports the private address of the client (inside the NAT).

Original comment by disco...@gmail.com on 3 Feb 2010 at 3:00

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same issue on Droid 2.1
I cannot hear the other party - tried from 3G and WLAN.
Sometimes enabling the speaker during the dialing phase solves this issue. Most 
of the 
time though the dialing phase stops after about 10 seconds without signaling 
even a 
calling tone.

Original comment by sakel...@gmail.com on 21 Feb 2010 at 6:45

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sorry, forgot to mention that when this condition exists then using the volume 
rocker 
invokes the ringer tone volume instead of media volume. It looks like the phone 
is not 
routing the voip/media audio correctly

Original comment by sakel...@gmail.com on 21 Feb 2010 at 6:49