transmitting, but not receiving

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transmitting, but not receiving

Doug Smith-6

To whom it may concern:

My name is Doug Smith and I am working on including linphone into the
next release of the Oralux Linux distribution for the Blind.  I am a
member of the development team, and I am currently testing linphone
and I have a probvlem with transmission of voice data to the remote
end.  

I can hear the remote party perfectly, but I have not been able to be
heard by them.  I have, here, included a level 5 debugging session
with linphone in an attempt to connect to:

sip:[hidden email]

where I have an account.  

I hope to use this program to help other Linux users in real-time.
This is one of the final hurdles I need to overcome before I put up my
web site for technical support and Linux software development.  Here
is the debug file I have.  

oRTP:(GLogLevel=32)** oRTP-0.7.1initialized.
MediaStreamer:(GLogLevel=32)** Found /dev/dsp.
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is 66.169.86.132.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is 66.169.86.132.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is 66.169.86.132.

| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 1 1739435548
| INFO2 | <ict.c: 35> allocating ICT context
| INFO2 | <eXutils.c: 511> Not an IPv4 or IPv6 address: fwd.pulver.com
| INFO2 | <eXutils.c: 541> DNS resolution with fwd.pulver.com:5060
| INFO1 | <jcallback.c: 148> Message sent:
INVITE sip:[hidden email] SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK211929910
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Subject: Phone call
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:   357

v=0
o=knoppix 123456 654321 IN IP4 66.169.86.132
s=A conversation
c=IN IP4 66.169.86.132
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 515> cb_sndinvite (id=1)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:457403!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
LinphoneCore:(GLogLevel=32)** CALL_RINGING

LinphoneCore:(GLogLevel=32)** Remote ringing...
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: ringplay,0 -> OssWrite,0
MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound System)] in playback mode with stereo=0,rate=8000,bits=16
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
MediaStreamer:(GLogLevel=32)** dsp blocksize is 2048.
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
Record-Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 11782 11782 IN IP4 69.90.168.13
s=session
c=IN IP4 69.90.168.13
t=0 0
m=audio 17896 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=1)
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is 66.169.86.132.

| INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
| INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
| INFO1 | <jcallback.c: 148> Message sent:
ACK sip:613@69.90.168.13:5028 SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1382328653
Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 20 ACK
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
LinphoneCore:(GLogLevel=32)** CALL_ANSWERED

MediaStreamer:(GLogLevel=32)** Mediastreamer processing thread is exiting.
MediaStreamer:(GLogLevel=32)** Closing writing channel of soundcard.
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: OssRead,0 -> MULAWEncoder,0
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: MULAWEncoder,0 -> RTPSend,0
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: RTPRecv,0 -> MULAWDecoder,0
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: MULAWDecoder,0 -> OssWrite,0
MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound System)] in capture mode with stereo=0,rate=8000,bits=16
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
MediaStreamer:(GLogLevel=32)** dsp blocksize is 2048.
MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound System)] in playback mode with stereo=0,rate=8000,bits=16
LinphoneCore:(GLogLevel=32)** CALL_STARTAUDIO

| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is 66.169.86.132.

| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 2 1739435548
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
| INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
| INFO1 | <jcallback.c: 148> Message sent:
BYE sip:613@69.90.168.13:5028 SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK6617564
Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 21 BYE
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 530> cb_sndbye (id=2)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:495101!
MediaStreamer:(GLogLevel=32)** Mediastreamer processing thread is exiting.
MediaStreamer:(GLogLevel=32)** Closing reading channel of soundcard.
MediaStreamer:(GLogLevel=32)** Closing writing channel of soundcard.
oRTP-stats:(GLogLevel=32)**
   Global statistics :
 number of rtp packet sent=0
 number of rtp bytes sent=0 bytes
 number of rtp packet received=784
 number of rtp bytes received=134848 bytes
 number of incoming rtp bytes successfully delivered to the application=134676
 number of times the application queried a packet that didn't exist=1424
 number of rtp packets received too late=0
 number of rtp packets skipped=1
 number of bad formatted rtp packets=0
 number of packet discarded because of queue overflow=0

| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK6617564
Record-Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 21 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=2)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:999923!
| INFO1 | <eXosip.c: 281> Release a terminated transaction
|  BUG  | <osip_transaction.c: 263> transaction already removed from list 1!
| INFO2 | <osip_transaction.c: 286> free transaction ressource 1 1739435548
| INFO2 | <ict.c: 112> free ict ressource
| INFO2 | <osip_transaction.c: 286> free transaction ressource 2 1739435548
| INFO2 | <nict.c: 110> free nict ressource

There is a line in this where the v is set to 0.  I presume that this
is the volume setting for my outgoing message.  For some reason, I am
not able to receive back an echo from the server on the echo test
which results from calling the above sip url.  

Any information on this would be greatly appreciated.  



Thank you very much for any help you can give me.

--
Doug Smith: C.S.F.C.
Computer Scientist For CHRIST!
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Re: transmitting, but not receiving

Simon Morlat

I'm sorry I did not see your second message.
The log you sent me confirm that it 's certainly the oss driver bug. Please
try with the alsa driver.
Thanks

Simon

Le Vendredi 4 Novembre 2005 04:23, Doug Smith a écrit :
> ting for my outgoing message.  For some reason, I am
> not able to receive back an echo from the server on the echo test
> which results from calling the above sip url.  
>
> Any information on this would be greatly appreciated.  


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