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:[EMAIL PROTECTED] 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:[EMAIL PROTECTED] reg_identity=sip:[EMAIL PROTECTED] reg_expires=600 reg_sendregister=1 publish=0 [auth_info_0] username=zloty passwd=xxx realm="asterisk" -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zloty Sent: Monday, November 28, 2005 1:43 AM To: [email protected] 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 _______________________________________________ Linphone-users mailing list [email protected] http://lists.nongnu.org/mailman/listinfo/linphone-users _______________________________________________ Linphone-users mailing list [email protected] http://lists.nongnu.org/mailman/listinfo/linphone-users
