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

[Jon Mansey]

If I call my Asterisk box from X-pro under windows, it works fine, if I call
from Linphone I get this echo/feedback looping noise. I don't think its
acoustic feedback, its sounds like the tx is hardwired to the rx and there's
only one buffer worth of data passing.

I suspect my ALSA driver or soundcard, but have no way to confirm.

Thx,

jm



-----Original Message-----
From: Simon MORLAT [mailto:address@hidden 
Sent: Monday, November 24, 2003 3:04 AM
To: Jon Mansey
Cc: address@hidden
Subject: Re: [Linphone-users] 0.12.1 and audio looping/feeding back


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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