Hi folks,
I'm trying to deal with Sofia SIP,
I compiled all the necessary from source code.
I can't understand what is wrong since sofsip_cli seems to work for the
SIP part but it doesn't work for RTP audio.
I mean that with "-i gstreamer" sometimes my voice is received but very
very low and other side audio (both real voice and music-on-hold) are
received with 15 sec delay and bad quality.
With "-i fsgst" my voice from cli is received quite well (but still low)
and I can't hear any audio.
"-i farsight" doesn't work at all (but it's well know in the README)
GStreamer works well for me with audio and video, I can't understand
what I am doing wrong.
This is what I compiled:
sofia-sip-1.12.0
sofsip-cli-0.10.1
jthread-1.2.1
jrtplib-3.6.0
gstreamer-0.10.8
farsight
gst-plugins-farsight
gst-plugins-base-0.10.8
gst-plugins-bad-0.10.3
gst-plugins-good-0.10.3
gst-plugins-ugly-0.10.3
gst-ffmpeg-0.10.1
Any suggestion?
Thanks,
Massimo
begin:vcard
fn:Massimo Mazzeo
n:Mazzeo;Massimo
org:Voismart - Espia SRL
adr:;;Via Pinturicchio, 1;Milano;;20133;Italy
email;internet:[EMAIL PROTECTED]
title:R&D Engineer
tel;work:+39 02 70633354
tel;fax:+39 02 45487890
x-mozilla-html:FALSE
url:http://www.voismart.it/
version:2.1
end:vcard
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