On Wed, 2006-06-28 at 15:42 +0200, Massimo Mazzeo Ocello wrote: > Hi folks,
Hi Massimo, like you I did some testing with sofsip-cli. What I'm missing in your post is: 1. Did you use sofsip-cli on both ends? (If not against what client did you do your testing?) 2. What kind of network did you use? (WLAN, cable based ethernet) 3. What kind of devices did you use? (ordinary PCs?) To answer my own questions: I used this setup: Nokia770 <-> WLAN <-> PC. On the PC side I used twinkle (0.7.1 on debian unstable) most of the time. While doing calls I recorded the network traffic with ethereal on the PC side. > 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. Same here. > 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. In my first case the stream Nokia770 -> PC was absolutely nice, but nothing of the stream PC -> Nokia770 could be heard. The ethereal logs revealed that RTP packets containing the audio have been correctly send in both directions! That means there has to be a problem in the receiving part of sofsip-cli's pipeline. > "-i farsight" doesn't work at all (but it's well know in the README) Yep :(. > GStreamer works well for me with audio and video How did you test that? Here are my pipelines for the commandline: Nokia770 -> PC: --- Nokia770: "gst-launch dsppcmsrc ! rtppcmapay ! udpsink host=192.168.2.1 port=9090" --- PC: "gst-launch udpsrc port=9090 ! rtppcmadepay ! alawdec ! alsasink device=plughw:1" (or plughw:0 for the first sound card) --- result: A very short piece of audio can be heard on the PC which is followed by silence. ethereal however indicates from the captured traffic that PCM a-law audio in RTP was transmitted for the whole time, the sending pipeline was running. The receiving pipeline on the PC used the debian unstable version for all the gstreamer related parts. --- PC -> Nokia770: --- Nokia770: "gst-launch udpsrc port=9090 ! rtppcmadepay ! dsppcmsink" --- PC: "gst-launch alsasrc ! alawenc ! rtpg711pay ! udpsink host=192.168.2.15 port=9090" --- result: nice audio volume but a 3 second delay --- Maybe the latter can be tweaked somehow to remove the delay? I very much appreciate further pipeline setups/configurations to test RTP transmission of audio between a Nokia770 and a PC. I think it would be usefull to collect such test on the sofsip-cli Wiki pages. @Kai: It would be very nice if you could delegate my findings to the person at Nokia responsible for the gstreamer elements in maemo 2.0. I would like to know how and which pipelines of gstreamer elements can be successfully used for transmitting audio from the build in mic via RTP on the 770. > I can't understand what I am doing wrong. I don't think, you are the one doing something wrong. Most probably the problem is in the used gstreamer elements, that build up the pipelines for the audio communication (RTP packet stream creation is part of that, too). The most obvious indication for that assumption are the ethereal sniffs. > 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 The build of sofsip-cli for the 770 was done in the current version of maemo2.0 with the following third party libs: jthread-1.2.0 jrtplib-3.5.2 sofia-sip-1.12.0 sofsip-cli-0.10.1 (modified pipelines for fsgst) gst-plugins-farsight (20060530225357) (dpkg -l "gst*"| grep ^i) ii gstreamer-tools 0.10.4-osso6 ii gstreamer0.10-gnomevfs 0.10.5-osso7 ii gstreamer0.10-plugins-base 0.10.5-osso7 ii gstreamer0.10-plugins-good 0.10.2-osso11-1 ii gstreamer0.10-tools 0.10.4-osso6 ii libgstreamer-plugins-base0.10-0 0.10.5-osso7 ii libgstreamer-plugins-base0.10-dev 0.10.5-osso7 ii libgstreamer0.10-0 0.10.4-osso6 ii libgstreamer0.10-dev 0.10.4-osso6 Greets, Jonek. PS: How can sofsip-cli bound to only one specified network interface (SIP and RTP)? It would be nice to have an option to select the number of the ALSA soundcard to be used. PPS: If anybody else here is interested in testing sofsip-cli on the Nokia770, please contact me - I would like to share the source and binaries of my early port. Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 _______________________________________________ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
