Re: [Linphone-users]Speech path could not be through...

linphone-users

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

Re: [Linphone-users]Speech path could not be through...

From:	Kanika Garg
Subject:	Re: [Linphone-users]Speech path could not be through...
Date:	Tue, 21 Jan 2003 21:23:31 -0800 (PST)

Hi,

I also think that the problem might be of the audio
driver. As I am new to Linux, I am not able to trace
the fault. 
I am hereby attaching the trace of the linphone on
both the pc's as:
Linphone-pc#.trace

Also attached the result of /sbin/lsmod as:
lsmod-pc.output

Regards,
Kanika Garg

--- Simon Morlat <address@hidden> wrote:
> Hi,
> 
> I'm pretty sure that the problem you have is caused
> of a bug in the
> audio driver.
> First, send me the result of /sbin/lsmod for both
> machines so that I
> check that you are not using a buggy driver.
> If this is the case, then you will have to replace
> them by alsa-drivers
> http://www.alsa-project.org
> Simon
> 
> Le mar 21/01/2003 à 06:52, Kanika Garg a écrit :
> > Hi all,
> > 
> > I am facing some problem while putting a call
> through
> > using two instances of Linphone running on 2
> different
> > PCs on LAN.
> > 
> > The parameters that have been set are -
> > 
> > PC 1
> > ----
> > 
> > RTP port - 7000
> > SIP port - 5060
> > URL - sip:address@hidden
> > 
> > 
> > PC 2
> > ----
> > 
> > RTP port - 7000
> > SIP port - 5060
> > URL - sip:address@hidden
> > 
> > The interface used was eth0.  I selected speex
> codec
> > and audio input was mic. And I did not select the
> > checkbox to kill sound deamons, as specified in
> the
> > Linphone's user manual. When I initiate a call
> from PC
> > 1, it shows in the status bar, 
> > 
> > Contacting <sip:address@hidden:5060> 
> > 
> > and on the other PC, it shows that - 
> > sip:address@hidden:5060 is calling you..... 
> > 
> > and I am able to hear the ring tone. But when the
> > incoming call is answered on PC 2, speech path is
> not
> > put through. I can only hear some sound, some
> noise
> > and not what is being said on the mic, on the
> other
> > hand.  Can anyone help me out in this regard... If
> > required, I can attach the output/trace.
> > 
> > Thanking you,
> > 
> > Regards,
> > Kanika Garg
> > HFCL, R&D Division,
> > Gurgaon
> > 
> > 
> > __________________________________________________
> > Do you Yahoo!?
> > Yahoo! Mail Plus - Powerful. Affordable. Sign up
> now.
> > http://mailplus.yahoo.com
> > 
> > 
> > _______________________________________________
> > Linphone-users mailing list
> > address@hidden
> >
>
http://mail.nongnu.org/mailman/listinfo/linphone-users
> -- 
> Simon Morlat <address@hidden>
> 

__________________________________________________
Do you Yahoo!?
Yahoo! Mail Plus - Powerful. Affordable. Sign up now.
http://mailplus.yahoo.com

__________________________________________________
Do you Yahoo!?
Yahoo! Mail Plus - Powerful. Affordable. Sign up now.
http://mailplus.yahoo.com

1) Starting linphone


address@hidden /root]# linphone
| INFO1 | <osipua.c: 59> Starting osip stack and osipua layer.

| INFO1 | <udp.c: 76> Entering osipua thread.

MediaStreamer-Message: Detected /dev/dsp -

MediaStreamer-WARNING **: Unable to find mixer for id 1.
MediaStreamer-Message: Detected /dev/dsp1 -

MediaStreamer-WARNING **: Unable to find mixer for id 3.
Found 2 interfaces.
Found eth0 interface with ip address 192.168.8.70

GnomeUI-WARNING **: Could not open help topics file NULL
Adding <sip:address@hidden> to the list of alias.
Message: Added audio payload type to SDP properties 110 speex-4/8000/1
Message: Added audio payload type to SDP properties 8 PCMA/8000/1
Message: Added audio payload type to SDP properties 0 PCMU/8000/1
Message: Added audio payload type to SDP properties 3 gsm/8000/1
Message: Added audio payload type to SDP properties 115 lpc10-1.5/8000/1
Message: Added audio payload type to SDP properties 111 speex_lbr-4/8000/1
MediaStreamer-Message: dsp blocksize is 512.


2) After sending a call request

state=0
invite state=1
| INFO1 | <udp.c: 191> Sending message:
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Type: application/sdp
Content-Length: 357

v=0
o=kanika 123456 654321 IN IP4 192.168.8.70
s=A conversation
c=IN IP4 192.168.8.70
t=0 0
m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
a=rtpmap:110 speex-4/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:115 lpc10-1.5/8000/1
a=rtpmap:111 speex_lbr-4/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 191> Sending message:
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Type: application/sdp
Content-Length: 357

v=0
o=kanika 123456 654321 IN IP4 192.168.8.70
s=A conversation
c=IN IP4 192.168.8.70
t=0 0
m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
a=rtpmap:110 speex-4/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:115 lpc10-1.5/8000/1
a=rtpmap:111 speex_lbr-4/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Content-Length: 0



| INFO1 | <ict_callbacks.c: 41> OnEvent_New_Incoming1xxResponse!

| INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
Content-Length: 0



| INFO1 | <ict_callbacks.c: 41> OnEvent_New_Incoming1xxResponse!


3) On accepting request by the remote

MediaStreamer-Message: dsp blocksize is 512.
MediaStreamer-Message: ms_filter_add_link: ringplay,0 -> OssWrite,0
MediaStreamer-Message: dsp blocksize is 512. 
| INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 302

v=0
o=saurabh 123456 654321 IN IP4 192.168.8.65
s=A conversation
c=IN IP4 192.168.8.65
t=0 0
m=audio 7000 RTP/AVP 110 8 0 3 115 111
a=rtpmap:110 speex-4/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:115 lpc10-1.5/8000/1
a=rtpmap:111 speex_lbr-4/8000/1


| INFO1 | <ict_callbacks.c: 71> OnEvent_New_Incoming2xxResponse!

| INFO1 | <ict_callbacks.c: 122> Found body application/sdp

MediaStreamer-Message: dsp blocksize is 512.
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> SpeexEncoder,0
MediaStreamer-Message: ms_filter_add_link: SpeexEncoder,0 -> RTPSend,0
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> SpeexDecoder,0
MediaStreamer-Message: ms_filter_add_link: SpeexDecoder,0 -> OssWrite,0
MediaStreamer-Message: dsp blocksize is 512.
| INFO1 | <udp.c: 191> Sending message:
ACK sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK1250473180
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 20 ACK
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Length: 0


4) On releasing connection locally


| INFO1 | <ict_callbacks.c: 30> Transaction 1 killed.

state end=2
oRTP-stats-Message:
   Global statistics :
 packet_sent=7222
 sent=361100 bytes
 packet_recv=14447
 hw_recv=722350 bytes
 recv=270400 bytes
 unavaillable=7223 bytes
 outoftime=0
 bad=0
 discarded=181150

| INFO1 | <udp.c: 191> Sending message:
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 21 BYE
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Length: 0


| INFO1 | <udp.c: 191> Sending message:
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 21 BYE
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Length: 0


| INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 21 BYE
Content-Length: 0



| INFO1 | <nict_callbacks.c: 30> Transaction 2 killed.

| INFO1 | <osipdialog.c: 1599> Call leg is removed. It remains 0 call legs in 
the ua list.

Linphone 2
-----------


1) Messages after starting Linphone:


| INFO1 | <osipua.c: 59> Starting osip stack and osipua layer.

MediaStreamer-Message: Detected /dev/dsp -

MediaStreamer-WARNING **: Unable to find mixer for id 1.
MediaStreamer-Message: Detected /dev/dsp1 -

MediaStreamer-WARNING **: Unable to find mixer for id 3.
| INFO1 | <udp.c: 76> Entering osipua thread.

Found 2 interfaces.
Found eth0 interface with ip address 192.168.8.65

GnomeUI-WARNING **: Could not open help topics file NULL
Adding <sip:address@hidden> to the list of alias.
Message: Added audio payload type to SDP properties 110 speex-4/8000/1
Message: Added audio payload type to SDP properties 0 PCMU/8000/1
Message: Added audio payload type to SDP properties 3 gsm/8000/1
Message: Added audio payload type to SDP properties 115 lpc10-1.5/8000/1
Message: Added audio payload type to SDP properties 8 PCMA/8000/1
Message: Added audio payload type to SDP properties 111 speex_lbr-4/8000/1

MediaStreamer-WARNING **: dsp block size set to 1024.
MediaStreamer-Message: dsp blocksize is 1024.




2) Message after receiving a call request:


| INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Type: application/sdp
Content-Length: 357

v=0
o=kanika 123456 654321 IN IP4 192.168.8.70
s=A conversation
c=IN IP4 192.168.8.70
t=0 0
m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
a=rtpmap:110 speex-4/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:115 lpc10-1.5/8000/1
a=rtpmap:111 speex_lbr-4/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11


| INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Type: application/sdp
Content-Length: 357

v=0
o=kanika 123456 654321 IN IP4 192.168.8.70
s=A conversation
c=IN IP4 192.168.8.70
t=0 0
m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
a=rtpmap:110 speex-4/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:115 lpc10-1.5/8000/1
a=rtpmap:111 speex_lbr-4/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11


| INFO1 | <ist_callbacks.c: 195> OnEvent_New_IncomingInvite!

| INFO1 | <ist_callbacks.c: 46> Sending 100 trying.

|WARNING| <uatransaction.c: 308> ua_transaction_execute: could not get dialog 
transaction.

| INFO1 | <osipua.c: 532> osip_ua_find 1: 192.168.8.65 <> 192.168.8.65

| INFO1 | <osipdialog.c: 157> <sip:address@hidden>;tag=609359131 has called at 
1043055721.

| INFO1 | <ist_callbacks.c: 123> Found body application/sdp.

| INFO1 | <ist_callbacks.c: 134> Creating a new body context.

| INFO1 | <sdpcontext.c: 126> sdp_context_notify_inc_req: negociation returned: 
200


MediaStreamer-WARNING **: dsp block size set to 1024.
MediaStreamer-Message: dsp blocksize is 1024.
MediaStreamer-Message: ms_filter_add_link: ringplay,0 -> OssWrite,0

MediaStreamer-WARNING **: dsp block size set to 1024.
MediaStreamer-Message: dsp blocksize is 1024.
invite handler done.
| INFO1 | <udp.c: 191> Sending message:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Content-Length: 0


| INFO1 | <udp.c: 191> Sending message:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
Content-Length: 0



3) On answering the call:

state=3

MediaStreamer-WARNING **: dsp block size set to 1024.
MediaStreamer-Message: dsp blocksize is 1024.
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> SpeexEncoder,0
MediaStreamer-Message: ms_filter_add_link: SpeexEncoder,0 -> RTPSend,0
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> SpeexDecoder,0
MediaStreamer-Message: ms_filter_add_link: SpeexDecoder,0 -> OssWrite,0

MediaStreamer-WARNING **: dsp block size set to 1024.
MediaStreamer-Message: dsp blocksize is 1024.
| INFO1 | <udp.c: 191> Sending message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 302

v=0
o=saurabh 123456 654321 IN IP4 192.168.8.65
s=A conversation
c=IN IP4 192.168.8.65
t=0 0
m=audio 7000 RTP/AVP 110 8 0 3 115 111
a=rtpmap:110 speex-4/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:115 lpc10-1.5/8000/1
a=rtpmap:111 speex_lbr-4/8000/1

| INFO1 | <ist_callbacks.c: 32> Transaction 1 killed.

invite state=4

oRTP-WARNING **: Error sending rtp packet: Connection refused.
| INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
ACK sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK1250473180
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 20 ACK
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Length: 0




oRTP-WARNING **: Error sending rtp packet: Connection refused.

MediaStreamer-WARNING **: MSTimer: must catchup 3 ticks.

MediaStreamer-WARNING **: MSTimer: must catchup 1 ticks.

MediaStreamer-WARNING **: MSTimer: must catchup 1 ticks.






4) After disconnecting from remote:


| INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 21 BYE
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Length: 0




oRTP-WARNING **: Error sending rtp packet: Connection refused.

oRTP-WARNING **: Error sending rtp packet: Connection refused.

oRTP-WARNING **: Error sending rtp packet: Connection refused.
| INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 21 BYE
max-forwards: 10
user-agent: oSIP/Linphone-0.9.1
Content-Length: 0



| INFO1 | <nist_callbacks.c: 62> nist_bye_received():

| INFO1 | <osipdialog.c: 1158> call-leg has been found!

oRTP-stats-Message:
   Global statistics :
 packet_sent=14662
 sent=733100 bytes
 packet_recv=7222
 hw_recv=361100 bytes
 recv=361100 bytes
 unavaillable=7325 bytes
 outoftime=0
 bad=0
 discarded=0

bye handler: state=0
| INFO1 | <udp.c: 191> Sending message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
From: <sip:address@hidden>;tag=609359131
To: <sip:address@hidden>;tag=2119830184
Call-ID: address@hidden
CSeq: 21 BYE
Content-Length: 0

PC-1 lsmod output
------------------

Module                  Size  Used by
via82cxxx_audio        17552   1  (autoclean)
ac97_codec              8800   0  (autoclean) [via82cxxx_audio]
soundcore               4464   2  (autoclean) [via82cxxx_audio]
autofs                 11264   1  (autoclean)
8139too                16480   1  (autoclean)
ipchains               38976   0  (unused)
usb-uhci               20720   0  (unused)
usbcore                49664   1  [usb-uhci]

--------------------------------------------------------------------

PC-2 lsmod output
-------------------

Module                  Size  Used by    Not tainted
via82cxxx_audio        20448   1  (autoclean)
uart401                 7936   0  (autoclean) [via82cxxx_audio]
ac97_codec             11904   0  (autoclean) [via82cxxx_audio]
sound                  72012   0  (autoclean) [via82cxxx_audio uart401]
soundcore               6692   4  (autoclean) [via82cxxx_audio sound]
binfmt_misc             7556   1 
nfsd                   76160   8  (autoclean)
lockd                  56736   1  (autoclean) [nfsd]
sunrpc                 75764   1  (autoclean) [nfsd lockd]
autofs                 12164   0  (autoclean) (unused)
eepro100               20336   1 
ipchains               43560  10 
ide-cd                 30272   0  (autoclean)
cdrom                  32192   0  (autoclean) [ide-cd]
nls_iso8859-1           3488   3  (autoclean)
nls_cp437               5120   3  (autoclean)
vfat                   12092   3  (autoclean)
fat                    37400   0  (autoclean) [vfat]
ext3                   67136   1 
jbd                    49400   1  [ext3]

[Prev in Thread]

Current Thread

[Next in Thread]

[Linphone-users]Speech path could not be through..., Kanika Garg, 2003/01/21
- Re: [Linphone-users]Speech path could not be through..., Simon Morlat, 2003/01/21
  - Re: [Linphone-users]Speech path could not be through..., Kanika Garg <=
    - Re: [Linphone-users]Speech path could not be through..., Simon Morlat, 2003/01/22

Prev by Date: Re: [Linphone-users]Speech path could not be through...
Next by Date: Re: [Linphone-users]Speech path could not be through...
Previous by thread: Re: [Linphone-users]Speech path could not be through...
Next by thread: Re: [Linphone-users]Speech path could not be through...
Index(es):
- Date
- Thread