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No audio in second session and freeze on end call #159

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?
1. A (imsdroid) initiates call to C
2. C (imsdroid) answers 
3. B (xlite) initiates call to C
4. C (imsdroid) answers 

What is the expected output? What do you see instead?
C should hear remote party after answering call from B

What version of the product are you using? On what operating system?
IMSdroid .357, latest build from svn, only A-law codec enabled  
On two Nexus-I's running Android 2.2
X-Lite on Windows as a third endpoint
Asterisk VOIP gateway.

Please provide any additional information below.
On answering the second call we see in logcat:

A phone1 = 701
C phone2 = 702
B XLite = 703

01-19 14:38:59.989: ERROR/Sensors(88): CAPELLA_CM3602_IOCTL_ENABLE error (I/O 
error)
01-19 14:39:00.139: DEBUG/org.doubango.imsdroid.Screens.ScreenAV(1309): Can 
detect orientation
01-19 14:39:06.009: 
DEBUG/org.doubango.imsdroid.media.MyProxyAudioConsumer(1309): 
prepareCallback(20,8000,1)
01-19 14:39:06.020: 
DEBUG/org.doubango.imsdroid.media.MyProxyAudioConsumer(1309): startCallback
01-19 14:39:06.020: 
DEBUG/org.doubango.imsdroid.media.MyProxyAudioProducer(1309): 
prepareCallback(20,8000,1)
01-19 14:39:06.020: 
DEBUG/org.doubango.imsdroid.media.MyProxyAudioConsumer(1309): ===== Audio 
Player Thread (Start) =====
01-19 14:39:07.460: ERROR/AudioHardwareQSD(59): Cannot open /dev/msm_pcm_in 
errno: 16
01-19 14:39:07.460: INFO/AudioHardwareQSD(59): do input routing device 40000
01-19 14:39:07.460: INFO/AudioHardwareQSD(59): Routing audio to Handset
01-19 14:39:07.599: DEBUG/AudioHardwareQSD(59): Switching audio device to 
01-19 14:39:07.599: DEBUG/AudioHardwareQSD(59): Handset
01-19 14:39:07.700: INFO/AudioHardwareQSD(59): AudioHardware PCM record is 
going to standby.
01-19 14:39:07.700: ERROR/AudioRecord(1309): Could not get audio input for 
record source 1
01-19 14:39:07.700: ERROR/AudioRecord-JNI(1309): Error creating AudioRecord 
instance: initialization check failed.
01-19 14:39:07.700: ERROR/AudioRecord-Java(1309): [ android.media.AudioRecord ] 
Error code -20 when initializing native AudioRecord object.
01-19 14:39:07.700: 
ERROR/org.doubango.imsdroid.media.MyProxyAudioProducer(1309): prepare() failed

Furthermore when C drops the call from A and the drops the call from B the 
display gets stuck in the 'Ending call' screen while the timer is still ticking 
(but the call is properly ended on A and B).

Original issue reported on code.google.com by theba...@gmail.com on 21 Jan 2011 at 5:45

GoogleCodeExporter commented 9 years ago 
On a related test we offered a call to the party which initated the call in the 
first session (instead of answered). 

What steps will reproduce the problem?
1. A (imsdroid) initiates call to C
2. C (imsdroid) answers 
3. B (xlite) initiates call to A
4. A (imsdroid) answers 

And found this in the log:
01-19 14:05:32.830: ERROR/ActivityManager(88): ANR in org.doubango.imsdroid 
(org.doubango.imsdroid/.Main)
01-19 14:05:32.830: ERROR/ActivityManager(88): Reason: keyDispatchingTimedOut
01-19 14:05:32.830: ERROR/ActivityManager(88): Load: 2.28 / 1.46 / 1.14
01-19 14:05:32.830: ERROR/ActivityManager(88): CPU usage from 32831ms to 14ms 
ago:
01-19 14:05:32.830: ERROR/ActivityManager(88):   mediaserver: 11% = 10% user + 
0% kernel / faults: 67 minor 68 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   system_server: 4% = 2% user + 
1% kernel / faults: 701 minor 18 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   ubango.imsdroid: 3% = 2% user 
+ 1% kernel / faults: 242 minor 16 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   akmd: 1% = 0% user + 1% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   dhd_dpc: 1% = 0% user + 1% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   putmethod.latin: 0% = 0% user 
+ 0% kernel / faults: 75 minor 4 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   events/0: 0% = 0% user + 0% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   synaptics_wq: 0% = 0% user + 
0% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   wpa_supplicant: 0% = 0% user + 
0% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   ksoftirqd/0: 0% = 0% user + 0% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   d.process.media: 0% = 0% user 
+ 0% kernel / faults: 21 minor
01-19 14:05:32.830: ERROR/ActivityManager(88):   .quicksearchbox: 0% = 0% user 
+ 0% kernel / faults: 27 minor 1 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   logcat: 0% = 0% user + 0% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   servicemanager: 0% = 0% user + 
0% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   m.android.phone: 0% = 0% user 
+ 0% kernel / faults: 37 minor 1 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   ndroid.launcher: 0% = 0% user 
+ 0% kernel / faults: 23 minor 1 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   ndroid.settings: 0% = 0% user 
+ 0% kernel / faults: 20 minor
01-19 14:05:32.830: ERROR/ActivityManager(88):   .cooliris.media: 0% = 0% user 
+ 0% kernel / faults: 30 minor
01-19 14:05:32.830: ERROR/ActivityManager(88): TOTAL: 23% = 16% user + 6% 
kernel + 0% softirq
01-19 14:05:52.000: ERROR/AudioRecord(433): Could not get audio input for 
record source 1
01-19 14:05:52.000: ERROR/AudioRecord-JNI(433): Error creating AudioRecord 
instance: initialization check failed.
01-19 14:05:52.000: ERROR/AudioRecord-Java(433): [ android.media.AudioRecord ] 
Error code -20 when initializing native AudioRecord object.
01-19 14:05:52.000: 
ERROR/org.doubango.imsdroid.media.MyProxyAudioProducer(433): prepare() failed

Original comment by theba...@gmail.com on 21 Jan 2011 at 6:02

GoogleCodeExporter commented 9 years ago 
I asked Matthijs who ran the test above to reproduce it including the SIP 
traffic. Perhaps this is helpful in finding what's wrong.

A = 701 (imsdroid initiating first call)
B = 703 (xlite initiating second call)
C = 702 (imsdroid answering first call)

*===============* 701 calls 702 
INVITE sip:702@10.100.255.17 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK40e95279;rport
Max-Forwards: 70
From: "701" <sip:701@10.100.255.36>;tag=as07ca04b5
To: <sip:702@10.100.255.17>
Contact: <sip:701@10.100.255.36>
Call-ID: 0b2b7f425f30c62615b7d2fc4ca42226@10.100.255.36
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 14:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 1433827620 1433827620 IN IP4 10.100.255.36
s=Asterisk PBX 1.6.2.9
c=IN IP4 10.100.255.36
t=0 0
m=audio 16034 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.100.255.17:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK40e95279;rport=5060
From: "701" <sip:701@10.100.255.36>;tag=as07ca04b5
To: <sip:702@10.100.255.17>
Call-ID: 0b2b7f425f30c62615b7d2fc4ca42226@10.100.255.36
CSeq: 102 INVITE
Server: OpenSIPS (1.5.3-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.100.255.17:5060 --->
SIP/2.0 180 Ringing
From: "701"<sip:701@10.100.255.36>;tag=as07ca04b5
To: <sip:702@10.100.255.17>;tag=624967058
Contact: <sip:702@10.100.0.164:55688;transport=udp>
Call-ID: 0b2b7f425f30c62615b7d2fc4ca42226@10.100.255.36
CSeq: 102 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 
10.100.255.36:5060;rport=5060;received=10.100.255.36;branch=z9hG4bK40e95279
Record-Route: <sip:10.100.255.17;lr=on>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (10 headers 0 lines) ---

<--- Transmitting (no NAT) to 10.100.255.17:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
10.100.255.17;branch=z9hG4bK8f95.60c46b97.0;received=10.100.255.17
Via: SIP/2.0/UDP 
10.100.50.8:48710;received=10.100.50.8;branch=z9hG4bK62879168;rport=48710
Record-Route: <sip:10.100.255.17;lr=on>
From: <sip:701@10.100.255.17>;tag=1743636289
To: <sip:702@10.100.255.17>;tag=as4247cbc7
Call-ID: a499703b-6375-3067-9b78-a137aef4caba
CSeq: 769455052 INVITE
Server: Asterisk PBX 1.6.2.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:702@10.100.255.36>
Content-Length: 0

*===============* 702 picks up (ok)
<------------>

<--- SIP read from UDP:10.100.255.17:5060 --->
SIP/2.0 200 OK
From: "701"<sip:701@10.100.255.36>;tag=as07ca04b5
To: <sip:702@10.100.255.17>;tag=624967058
Contact: <sip:702@10.100.0.164:55688;transport=udp>
Call-ID: 0b2b7f425f30c62615b7d2fc4ca42226@10.100.255.36
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 202
Via: SIP/2.0/UDP 
10.100.255.36:5060;rport=5060;received=10.100.255.36;branch=z9hG4bK40e95279
Record-Route: <sip:10.100.255.17;lr=on>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=doubango 1983 678902 IN IP4 10.100.0.164
s=-
c=IN IP4 10.100.0.164
t=0 0
m=audio 7214 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 
(alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.100.0.164:7214
list_route: hop: <sip:10.100.255.17;lr=on>
set_destination: Parsing <sip:10.100.255.17;lr=on> for address/port to send to
set_destination: set destination to 10.100.255.17, port 5060
Transmitting (no NAT) to 10.100.255.17:5060:
ACK sip:702@10.100.0.164:55688;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK2968aa10;rport
Route: <sip:10.100.255.17;lr=on>
Max-Forwards: 70
From: "701" <sip:701@10.100.255.36>;tag=as07ca04b5
To: <sip:702@10.100.255.17>;tag=624967058
Contact: <sip:701@10.100.255.36>
Call-ID: 0b2b7f425f30c62615b7d2fc4ca42226@10.100.255.36
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9
Content-Length: 0

---
Audio is at 10.100.255.36 port 17530
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.100.255.17:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.100.255.17;branch=z9hG4bK8f95.60c46b97.0;received=10.100.255.17
Via: SIP/2.0/UDP 
10.100.50.8:48710;received=10.100.50.8;branch=z9hG4bK62879168;rport=48710
Record-Route: <sip:10.100.255.17;lr=on>
From: <sip:701@10.100.255.17>;tag=1743636289
To: <sip:702@10.100.255.17>;tag=as4247cbc7
Call-ID: a499703b-6375-3067-9b78-a137aef4caba
CSeq: 769455052 INVITE
Server: Asterisk PBX 1.6.2.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:702@10.100.255.36>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 826694848 826694848 IN IP4 10.100.255.36
s=Asterisk PBX 1.6.2.9
c=IN IP4 10.100.255.36
t=0 0
m=audio 17530 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:10.100.255.17:5060 --->
ACK sip:702@10.100.255.36 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.17;branch=z9hG4bK8f95.60c46b97.2
Via: SIP/2.0/UDP 
10.100.50.8:48710;received=10.100.50.8;branch=z9hG4bK2108429659;rport=48710
From: <sip:701@10.100.255.17>;tag=1743636289
To: <sip:702@10.100.255.17>;tag=as4247cbc7
Contact: 
<sip:701@10.100.50.8:48710;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp
.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Call-ID: a499703b-6375-3067-9b78-a137aef4caba
CSeq: 769455052 ACK
Content-Length: 0
Max-Forwards: 69
Proxy-Authorization: Digest 
username="701@10.100.255.17",realm="10.100.255.17",nonce="4d36fa94000002a89f99b7
0d61511a6f46ca7eabf04ad1a2",uri="sip:702@10.100.255.36",response="6721d8d024c892
192c2813b22456e2d0",algorithm=MD5
Accept-Contact:  *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
P-Preferred-Service:  urn:urn-7:3gpp-service.ims.icsi.mmtel
Allow: INVITE,ACK,CANCEL,BYE,MESSAGE,OPTIONS,NOTIFY,PRACK,UPDATE,REFER
Privacy: none
P-Access-Network-Info:  ADSL;utran-cell-id-3gpp=00000000
User-Agent: IM-client/OMA1.0 IMSDroid/v1.2.356 (doubango r545)

<------------->
--- (17 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK14f448aa;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as36b3c5dd
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 1cd85c3e57bd6fe406c064227b7615fd@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 14:51:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

*===============* 703(XLite) calls 701 (701 rings)
To: <sip:702@10.100.255.17>;tag=52830444
Contact: <sip:702@10.100.0.164:55688;transport=udp>
Call-ID: 3b9cff56486c28837a3c3ae5559687d9@10.100.255.36
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 203
Via: SIP/2.0/UDP 
10.100.255.36:5060;rport=5060;received=10.100.255.36;branch=z9hG4bK55521b38
Record-Route: <sip:10.100.255.17;lr=on>
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=doubango 1983 678902 IN IP4 10.100.0.164
s=-
c=IN IP4 10.100.0.164
t=0 0
m=audio 63282 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 
(alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.100.0.164:63282
list_route: hop: <sip:10.100.255.17;lr=on>
set_destination: Parsing <sip:10.100.255.17;lr=on> for address/port to send to
set_destination: set destination to 10.100.255.17, port 5060
Transmitting (no NAT) to 10.100.255.17:5060:
ACK sip:702@10.100.0.164:55688;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK6a5743b9;rport
Route: <sip:10.100.255.17;lr=on>
Max-Forwards: 70
From: "703" <sip:703@10.100.255.36>;tag=as265e2de3
To: <sip:702@10.100.255.17>;tag=52830444
Contact: <sip:703@10.100.255.36>
Call-ID: 3b9cff56486c28837a3c3ae5559687d9@10.100.255.36
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9
Content-Length: 0

---
Audio is at 10.100.255.36 port 12902
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 10.100.255.17:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.100.255.17;branch=z9hG4bK3721.41a1e307.0;received=10.100.255.17
Via: SIP/2.0/UDP 
172.17.119.85:42406;received=172.17.119.85;branch=z9hG4bK-d8754z-dc30847d5f236d0
3-1---d8754z-;rport=42406
Record-Route: <sip:10.100.255.17;lr=on>
From: "703"<sip:703@10.100.255.17>;tag=3d3b955b
To: "702"<sip:702@10.100.255.17>;tag=as545b336f
Call-ID: Nzg1OGZmMmJjMTRmZWM5ZDhiOTliYTcyMjJmZTNiZjg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:702@10.100.255.36>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 1804464039 1804464039 IN IP4 10.100.255.36
s=Asterisk PBX 1.6.2.9
c=IN IP4 10.100.255.36
t=0 0
m=audio 12902 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:10.100.255.17:5060 --->
ACK sip:702@10.100.255.36 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.17;branch=z9hG4bK3721.41a1e307.2
Via: SIP/2.0/UDP 
172.17.119.85:42406;received=172.17.119.85;branch=z9hG4bK-d8754z-4049171ce505057
0-1---d8754z-;rport=42406
Max-Forwards: 69
Contact: <sip:703@172.17.119.85:42406>
To: "702"<sip:702@10.100.255.17>;tag=as545b336f
From: "703"<sip:703@10.100.255.17>;tag=3d3b955b
Call-ID: Nzg1OGZmMmJjMTRmZWM5ZDhiOTliYTcyMjJmZTNiZjg.
CSeq: 2 ACK
Proxy-Authorization: Digest 
username="703",realm="10.100.255.17",nonce="4d36fd1a000002afa6214dfa8d1788ce2435
158e4ef1233b",uri="sip:702@10.100.255.17",response="1ae86a4f4da2637d56d2a7453122
b57d",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK519d7c99;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as1940a2da
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 478bb43401e7bd02702a89215ac63728@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:02:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '478bb43401e7bd02702a89215ac63728@10.100.255.36' 
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK6f7550cf;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as42ae1df2
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 2c00b7f40b69018c5dcb244b5db11fa6@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:02:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '2c00b7f40b69018c5dcb244b5db11fa6@10.100.255.36' 
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK272df46b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as581baefc
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 3a735a013869747b0fea049538b980b4@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:02:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '3a735a013869747b0fea049538b980b4@10.100.255.36' 
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK6770c4c3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as46a84c30
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 0160fddc48924ec118a6223e7a89e65f@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:02:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '0160fddc48924ec118a6223e7a89e65f@10.100.255.36' 
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK031247c2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as27137de6
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 75259ee95b4f0fca477eb15274dfae4f@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:03:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '75259ee95b4f0fca477eb15274dfae4f@10.100.255.36' 
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK5560c0d0;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as5491536a
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 3b202da2798208660bb17f016b034c64@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Really destroying SIP dialog '3b202da2798208660bb17f016b034c64@10.100.255.36' 
Method: OPTIONS
Reliably Transmitting (no NAT) to 10.100.255.21:5060:
OPTIONS sip:10.100.255.21 SIP/2.0
Via: SIP/2.0/UDP 10.100.255.36:5060;branch=z9hG4bK4893122e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.255.36>;tag=as6e14c881
To: <sip:10.100.255.21>
Contact: <sip:asterisk@10.100.255.36>
Call-ID: 7ef1bf2d6800503648dbcb316ccd4bb3@10.100.255.36
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9
Date: Wed, 19 Jan 2011 15:03:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

*===============* 701 presses pickup button
*===============* (703 is in call)
*===============* (701's pickup button turns yellow)
*===============* (errors show in logcat)
*===============* (dialog displays IMSDROID has crashed, press ok to shutdown)
*===============* (702 and 703 are still in call)

01-19 14:05:32.830: ERROR/ActivityManager(88): ANR in org.doubango.imsdroid 
(org.doubango.imsdroid/.Main)
01-19 14:05:32.830: ERROR/ActivityManager(88): Reason: keyDispatchingTimedOut
01-19 14:05:32.830: ERROR/ActivityManager(88): Load: 2.28 / 1.46 / 1.14
01-19 14:05:32.830: ERROR/ActivityManager(88): CPU usage from 32831ms to 14ms 
ago:
01-19 14:05:32.830: ERROR/ActivityManager(88):   mediaserver: 11% = 10% user + 
0% kernel / faults: 67 minor 68 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   system_server: 4% = 2% user + 
1% kernel / faults: 701 minor 18 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   ubango.imsdroid: 3% = 2% user 
+ 1% kernel / faults: 242 minor 16 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   akmd: 1% = 0% user + 1% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   dhd_dpc: 1% = 0% user + 1% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   putmethod.latin: 0% = 0% user 
+ 0% kernel / faults: 75 minor 4 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   events/0: 0% = 0% user + 0% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   synaptics_wq: 0% = 0% user + 
0% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   wpa_supplicant: 0% = 0% user + 
0% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   ksoftirqd/0: 0% = 0% user + 0% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   d.process.media: 0% = 0% user 
+ 0% kernel / faults: 21 minor
01-19 14:05:32.830: ERROR/ActivityManager(88):   .quicksearchbox: 0% = 0% user 
+ 0% kernel / faults: 27 minor 1 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   logcat: 0% = 0% user + 0% 
kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   servicemanager: 0% = 0% user + 
0% kernel
01-19 14:05:32.830: ERROR/ActivityManager(88):   m.android.phone: 0% = 0% user 
+ 0% kernel / faults: 37 minor 1 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   ndroid.launcher: 0% = 0% user 
+ 0% kernel / faults: 23 minor 1 major
01-19 14:05:32.830: ERROR/ActivityManager(88):   ndroid.settings: 0% = 0% user 
+ 0% kernel / faults: 20 minor
01-19 14:05:32.830: ERROR/ActivityManager(88):   .cooliris.media: 0% = 0% user 
+ 0% kernel / faults: 30 minor
01-19 14:05:32.830: ERROR/ActivityManager(88): TOTAL: 23% = 16% user + 6% 
kernel + 0% softirq

*===============* (And this one follows immediately)

01-19 14:05:52.000: ERROR/AudioRecord(433): Could not get audio input for 
record source 1
01-19 14:05:52.000: ERROR/AudioRecord-JNI(433): Error creating AudioRecord 
instance: initialization check failed.
01-19 14:05:52.000: ERROR/AudioRecord-Java(433): [ android.media.AudioRecord ] 
Error code -20 when initializing native AudioRecord object.
01-19 14:05:52.000: 
ERROR/org.doubango.imsdroid.media.MyProxyAudioProducer(433): prepare() failed

Original comment by theba...@gmail.com on 21 Jan 2011 at 6:13

GoogleCodeExporter commented 9 years ago 
Is it possible to have wireshark capture instead of plain text?

Original comment by boss...@yahoo.fr on 26 Jan 2011 at 10:54

GoogleCodeExporter commented 9 years ago 
I'll work on the wireshark, will keep you posted, Matthijs

Original comment by matthijs...@gmail.com on 1 Feb 2011 at 10:32

GoogleCodeExporter commented 9 years ago 
I ran a similar test:
1. A calls B
2. B answers and a normal audio session is made
3. A places B on hold
4. A calls C
5. C answers and what seems like a normal session is made
6. However, in this session, only C is sending RTP to A;A is not sending any 
RTP to C.
7. The SDP attributes for send/recv are identical to the session between A & B.
8. It seems that the audio doesn't go from A towards C

Original comment by sand...@yahoo.com on 7 Mar 2011 at 7:28

GoogleCodeExporter commented 9 years ago 
@sand...@yahoo.com
Please provide information about your device,android version, imsdroid release, 
... and if possible logs or network trace.

Original comment by boss...@yahoo.fr on 7 Mar 2011 at 8:14

GoogleCodeExporter commented 9 years ago 
The devices are LG Optimus One P500(based on MSM 7227 chipset from Qualcomm). 
The Android version is 2.2 and I used imsdroid revision 366 from the trunk. 
Here's the wireshark trace on C(where A(ip addr 191.168.1.253) has made a call 
to B(192.168.1.247) first, placed it on hold & then set up a call with 
C(192.168.1.244)). As you can see, C is sending RTP back to A, but A is not 
sending anything to C.

Original comment by sand...@yahoo.com on 7 Mar 2011 at 10:25

Attachments:

GoogleCodeExporter commented 9 years ago 
Forgot to add that the SIP proxy is running on 192.168.1.244, the same as 
C.Just so that doesn't throw you off.
-Sandeep

Original comment by sand...@yahoo.com on 7 Mar 2011 at 10:27

GoogleCodeExporter commented 9 years ago 
Fixed in 2.x

Original comment by boss...@yahoo.fr on 23 Jul 2011 at 5:03