mayank-kratin / libjingle

Automatically exported from code.google.com/p/libjingle
0 stars 0 forks source link

QOSAddSocketToFlow failed to add a flow with error 87 #357

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
What steps will reproduce the problem?
1. run call.exe -d 
2. make a voice call to a phone: call +14443339999@voice.google.com
3. program makes connection and I can send audio but I cannot receive any. The 
same code works in linux ubuntu. Looking at the output I saw an error related 
to  qwave.dll saying function QOSAddSocketToFlow returns with error 87. Does 
anybody know to what this error is related?

What is the expected output? What do you see instead?

This is the relevant output right before I see the message from the ortp:

....
Contents are not grouped together cannot be muxed
Session:4204145883 Old state:STATE_SENTINITIATE New state:STATE_RECEIVEDACCEPT 
Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting remote voice description
Using speex/16000
check OS support for qwave.lib: 6 1 7601
QOSAddSocketToFlow failed to add a flow with error 87

Failed to set DSCP value on socket.
speex_lib_ctl does not support SPEEX_LIB_CPU_FEATURE_NEON
Payload's bitrate is 28000
Setting audio encoder network bitrate to 28000
ms_filter_link: MSWinSndRead:00BAA7D0,0-->MSVolume:00BABF50,0
ms_filter_link: MSVolume:00BABF50,0-->MSSpeexEnc:00BAB890,0
ms_filter_link: MSSpeexEnc:00BAB890,0-->MSRtpSend:00BA8A28,0
ms_filter_link: MSRtpRecv:00BAA450,0-->MSSpeexDec:00BABAF0,0
ms_filter_link: MSSpeexDec:00BABAF0,0-->MSDtmfGen:00BAA5E0,0
ms_filter_link: MSDtmfGen:00BAA5E0,0-->MSVolume:00BAC1E8,0
ms_filter_link: MSVolume:00BAC1E8,0-->MSEqualizer:00BAA2B0,0
ms_filter_link: MSEqualizer:00BAA2B0,0-->MSWinSndWrite:00BAB030,0
win32 timer resolution set to 2 ms
Setting maxbitrate=12000 to speex encoder.
Using bitrate 9800 for speex encoder, ip bitrate is 25600
Exiting for with codec speex/16000
Add remote ssrc: 1888156779
SRTP activated with negotiated parameters: send cipher_suite 
AES_CM_128_HMAC_SHA1_80 recv cipher_suite AES_CM_128_HMAC_SHA1_80
SetSend Mute flag is 0
Changing voice state, recv=1 send=1
....

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

libjingle-0.6.14 and linphone, windows 7 64 bit. 

Please provide any additional information below.
I compiled mediastreamer2 and libjingle with visual studio 2010 and used 
linphonemediaengine.cc

Original issue reported on code.google.com by ame68...@gmail.com on 3 Jun 2012 at 9:22

GoogleCodeExporter commented 9 years ago
Anybody?

Original comment by ame68...@gmail.com on 7 Jun 2012 at 5:16

GoogleCodeExporter commented 9 years ago
Linphone is no longer supported.

Original comment by juberti@google.com on 31 May 2013 at 8:48