Crappy audio [i'm totally banging my head]

[GoogleCodeExporter]

What steps will reproduce the problem?
1. Installing .91 rc1 binary
2. Putting sip_s40 into webapps/sip
3. Registering to an asterisk server

What is the expected output? What do you see instead?
I run in debug mode. Am using flash player 10. I'm able to make a call, but 
audio is VERY inconsistent. Whenever i dial 4242(default voicemail app) i 
sometimes hear the greeting, sometimes i do not. Asterisk is fully working with 
other clients. Whenever i dial some of my phones and pick-up i can hear the 
webclient audio clearly on the ata, but ata audio is nowhere to be heard. In 
rare cases, i do get audio, but it's echoing and lousy. I fell back to 
ulaw/alaw in my sip.conf with that specific webuser. I do the test on the same 
subnet and firewalls are off.
tcpdumps seem ok, udp packets are flowing from the asterisk to the box, and 
1935(rtmp) packets are flowing from the red5box to the client.

What version of the product are you using? On what operating system?
using on ubuntu 9.04

Here're my debug logs :
SipUserAgent - onCallAccepted -> newSdp = v=0                                   

o=root 1554103232 1554103232 IN IP4 xxx.yyy.aaa.bbb                             

s=Asterisk PBX 1.6.1.6                                                          

c=IN IP4 127.0.1.1       (is this normal???)                                    

t=0 0                                                                           

m=audio 3008 RTP/AVP 0                                                          

a=rtpmap:0 PCMU/8000                                                            

a=rtpmap:8 PCMA/8000/1                                                          

a=rtpmap:18 G729/8000/1                                                         

a=fmtp:18 annexb=no                                                             

a=rtpmap:111 ILBC/8000/1                                                        

a=fmtp:111 mode=30                                                              

a=ptime:20                                                                      

.                                                                               

SIPCodecUtils - initSipAudioCodec -> Codec id = [0], codec name = [PCMU], 
sampleRate = [8000], incomingEndodedFrameSize = [160], incomingDedodedFrameSize 
= [160], incomingPacketization = [20], outgoingEndodedFrameSize = [160], 
outgoingDedodedFrameSize = [160], outgoingPacketization = [20].                 

[SIPUser] onUaCallConnected                                                     

SipUserAgent - launchMediaApplication -> localAudioPort = 3008, localVideoPort 
= 0.
SipUserAgent - launchMediaApplication -> remoteAudioPort = 17508, 
remoteVideoPort = 0.
SipUserAgent - launchMediaApplication -> user_profile.audio = true, 
user_profile.video = false, audio_app = null, video_app = null.
SipAudioLauncher - SIPAudioLauncher -> New audio sender to 
xxx.yyy.aaa.bbb:17508.
SipAudioLauncher - SIPAudioLauncher -> sender configs: payloadType = [0], 
payloadName = [PCMU].
SipAudioLauncher - SIPAudioLauncher -> New audio receiver on 3008.
SipAudioLauncher - startMedia -> Starting sip audio...
SipAudioLauncher - startMedia -> Start sending.
RTPStreamSender - start() -> using blocks of 160 bytes.
SipAudioLauncher - startMedia -> Start receiving.
[SIPUser] onUaCallAccepted
RtpStreamReceiver - run -> internalBuffer.length = 172, codedBuffer.length = 
160, decodingBuffer.length = 160.
[INFO] [NioProcessor-1] org.red5.server.net.rtmp.RTMPHandler - Remembering 
client buffer on stream: 0
[INFO] [NioProcessor-1] org.red5.server.stream.ClientBroadcastStream - Consumer 
connect
[INFO] [NioProcessor-1] org.red5.server.stream.ClientBroadcastStream - Provider 
connect
[INFO] [NioProcessor-1] org.red5.server.stream.ClientBroadcastStream - Stream 
start
[INFO] [NioProcessor-1] org.red5.server.stream.ClientBroadcastStream - Provider 
connect
[SIPUser] streamStatus start
*** 65
+++ Audio - ts: 0 length: 65
*** 65
*** 65
+++ Audio - ts: 64 length: 65
+++ Audio - ts: 128 length: 65
*** 65
*** 65
+++ Audio - ts: 192 length: 65
+++ Audio - ts: 256 length: 65
*** 65
+++ Audio - ts: 320 length: 65
*** 65
*** 65
+++ Audio - ts: 384 length: 65
*** 65
+++ Audio - ts: 448 length: 65
+++ Audio - ts: 512 length: 65

Original issue reported on code.google.com by ser...@gmail.com on 10 Nov 2010 at 10:35

[GoogleCodeExporter]

There's a change with my situation now. I've been madly ruling out reasons for 
the crappy audio. I deployed the red5phone+red5 on an outside host with no NAT 
to the asterisk. I noticed a BIG difference with browsers on the WIN platform. 
IE has been giving me considerably better audio with no echo. Chrome and 
firefox give better performance on the flex. I have modified the flex ui to go 
for autologin, which is not working in IE. 

CHANGE:
i installed red5phone r42. there're no visible delays till now. i haven't gone 
into extensive testing yet. i call a pstn - it's ok. i got dynamic sip clients 
on that same asterisk which are ok too. my configuration will be hardened. i 
will put rtmp through an stunnel and the sip/rtp via an openvpn tunnel. will 
keep u notified on how it behaves.

Original comment by ser...@gmail.com on 16 Nov 2010 at 2:07