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Re: [Linphone-users] 0.12.1 and audio looping/feeding back

From: Simon MORLAT
Subject: Re: [Linphone-users] 0.12.1 and audio looping/feeding back
Date: Mon, 24 Nov 2003 12:04:00 +0100
User-agent: Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.5) Gecko/20031105 Thunderbird/0.3 
Hello,
YOu are probably dealing with some echo.
What is the sound equipement of the remote UA ?
Prefer using headsets than microphone+speaker.
Simon

Jon Mansey wrote:


Hi

So I upgraded to 0.12.1 and the ALSA i810 8K problem appears to have
been fixed.

Now though, when I connect to my asterisk box, all  hear is a click or
chirp which then loops around and builds like feedback. Any ideas what
may cause this? Below is the console output of the session.

Thanks for tips etc. Still desperate for a linux SIP UA!

Jon



address@hidden:~$ linphone
| INFO1 | <osipua.c: 65> Starting osip stack and osipua layer.

| INFO1 | <udp.c: 112> Entering osipua thread.

MediaStreamer-Message: Found /dev/dsp.
MediaStreamer-Message: Found ALSA device: Intel 82801CA-ICH3
MediaStreamer-Message: alsa_set_params:  blocksize=512.
| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <utils.c: 409> Outgoing interface to reach 192.168.2.2 is
192.168.2.205.

| ERROR | <osipdialog.c: 1681> generating_request_out_of_dialog: setting
ua->ua_family=2 from localip 192.168.2.205

| INFO1 | <udp.c: 295> Sending message:
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK3829477616
From: <sip:address@hidden>;tag=2269386534
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length:   371

v=0
o=jonm 123456 654321 IN IP4 192.168.2.205
s=A conversation
c=IN IP4 192.168.2.205
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:110 20
b=AS:111 28
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

| INFO1 | <udp.c: 186> info: Message from 192.168.2.2:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK3829477616
From: <sip:address@hidden>;tag=2269386534
To: <sip:address@hidden>;tag=as4886439c
Call-ID: address@hidden
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Length: 0



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

| INFO1 | <udp.c: 186> info: Message from 192.168.2.2:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK3829477616
From: <sip:address@hidden>;tag=2269386534
To: <sip:address@hidden>;tag=as4886439c
Call-ID: address@hidden
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 24152 24152 IN IP4 192.168.2.2
s=session
c=IN IP4 192.168.2.2
t=0 0
m=audio 14158 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


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

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

MediaStreamer-Message: alsa_set_params:  blocksize=512.
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> GSMEncoder,0
MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
MediaStreamer-Message: ms_filter_add_link: GSMDecoder,0 -> OssWrite,0
MediaStreamer-Message: Opening sound card in capture mode with
stereo=0,rate=8000,bits=16
MediaStreamer-Message: alsa_set_params:  blocksize=512.
MediaStreamer-Message: Opening sound card in playback mode with
stereo=0,rate=8000,bits=16
MediaStreamer-Message: alsa_set_params:  blocksize=512.
| INFO1 | <udp.c: 295> Sending message:
ACK sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK3615041925
From: <sip:address@hidden>;tag=2269386534
To: <sip:address@hidden>;tag=as4886439c
Call-ID: address@hidden
CSeq: 20 ACK
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


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


(linphone:934): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone:934): MediaStreamer-WARNING **: alsa_card_write: Error writing
sound buffer (size=512):Resource temporarily unavailable

(linphone:934): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone:934): MediaStreamer-WARNING **: alsa_card_write: Error writing
sound buffer (size=512):Resource temporarily unavailable

(linphone:934): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone:934): MediaStreamer-WARNING **: alsa_card_write: Error writing
sound buffer (size=512):Resource temporarily unavailable

(linphone:934): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone:934): MediaStreamer-WARNING **: alsa_card_write: Error writing
sound buffer (size=512):Resource temporarily unavailable

(linphone:934): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone:934): MediaStreamer-WARNING **: alsa_card_write: Error writing
sound buffer (size=512):Resource temporarily unavailable
MediaStreamer-Message: Mediastreamer processing thread is exiting.
oRTP-stats-Message:
  Global statistics :
packet_sent=633
sent=28485 bytes
packet_recv=634
hw_recv=29165 bytes
recv=28940 bytes
unavaillable=634 bytes
outoftime=0
bad=0
discarded=0

| INFO1 | <udp.c: 295> Sending message:
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK397684538
From: <sip:address@hidden>;tag=2269386534
To: <sip:address@hidden>;tag=as4886439c
Call-ID: address@hidden
CSeq: 21 BYE
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


| INFO1 | <udp.c: 186> info: Message from 192.168.2.2:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.205:5060;branch=z9hG4bK397684538
From: <sip:address@hidden>;tag=2269386534
To: <sip:address@hidden>;tag=as4886439c
Call-ID: address@hidden
CSeq: 21 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Length: 0



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

| INFO1 | <osipdialog.c: 1915> Dialog is removed. It remains 0 dialog(s)
in the ua list.










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