RE: [Linphone-users] Incoming sound problem

[Zloty]

Hi 

I check something and on my configuration file is
audio_rtp_port=7078
video_rtp_port=9078
I suppose that those ports should be used for sound communication, then why
when I use tcpdump I see:
20:30:23.222466 10.1.1.230.16000 > 10.1.1.239.49154: udp 172 (DF)
20:30:23.238333 10.1.1.239.49154 > 10.1.1.230.10500: udp 172
(10.1.1.230 is redhat with linphone,10.1.1.239 is win xp)
Then I think it linphone use for outgoing 16000 port and or incoming 10500, 
I can't find this 10500 in any configuration file.
I run this test few times and always port for incoming were 10500,for
outgoing has been changed every time.

What can happened?

This is my .linphonec file
[net]
con_type=3
use_nat=0
if_name=eth0

[sip]
sip_port=15060
guess_hostname=1
contact=sip:address@hidden
use_info=0
use_ipv6=0
default_proxy=0

[rtp]
audio_rtp_port=7078
video_rtp_port=9078
audio_jitt_comp=60
video_jitt_comp=60

[sound]
dev_id=5
rec_lev=80
play_lev=80
source=m
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
remote_ring=/usr/share/sounds/linphone/ringback.wav
playback_dev_id=5
capture_dev_id=5

[video]
enabled=0
show_local=0

[audio_codec_0]
mime=PCMU
rate=8000
enabled=1

[audio_codec_1]
mime=GSM
rate=8000
enabled=1

[audio_codec_2]
mime=PCMA
rate=8000
enabled=1

[audio_codec_3]
mime=speex
rate=8000
enabled=1

[audio_codec_4]
mime=speex
rate=16000
enabled=1

[audio_codec_5]
mime=1015
rate=8000
enabled=1

[proxy_0]
reg_proxy=sip:address@hidden
reg_identity=sip:address@hidden
reg_expires=600
reg_sendregister=1
publish=0

[auth_info_0]
username=zloty
passwd=xxx
realm="asterisk"

-----Original Message-----
From: address@hidden
[mailto:address@hidden On Behalf
Of Zloty
Sent: Monday, November 28, 2005 1:43 AM
To: address@hidden
Subject: [Linphone-users] Incoming sound problem

Hi,

I just installed environment to test linphone, and I have problem with
sound.
I have two computers :
1. win xp with sjphone
2. red hat 9 (two sound cards), (asterisk 1.0.7 and linphone 1.0.1 (1.1.0) )


Both clients linphone and sjphone register in asterisk. I can make call from
sjphone to asterisk - ok
linphone to asterisk - ok
linphone to asterisk - I don't hear this which I tell on sjphone, 

Previous I'm think that it's problem with config file, I ad if_name=eth0, 
but it doesn't help.

I check both versions 1.0.1 and 1.1.0 and result are the same.

Any suggestions?

Anyway I've problem with libspeex and 1.1.0 configuration, the solution were
very simply, install libspeex from rpm not from source.

Regards
Robert




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