On 6/2/06, H. L. <[EMAIL PROTECTED]
> wrote:
Hi Kai:
Any clue why the basic gstreamer is still used in the following? I used sofsip-cli-0.10, farsight version is 0.1.3.1. But when I installed gst-plugins-farsight, I ran "make install", but I cannot run "gst-register"(command not found) as specified in README. I don't know if this is the problem.
thanks!On 6/1/06, H. L. < [EMAIL PROTECTED]> wrote:I used the same command, however it reported "selecting media impl: gstreamer", so RTP is still not symmetric. I don't know what's wrong.thanks a lot!
On 6/1/06, [EMAIL PROTECTED] < [EMAIL PROTECTED]> wrote:Hi,
to use gst-plugins-farsight, it should be enough to just install
it. Then start sofsip_cli with "./sofsip_cli --media-impl=fsgst" and
it should report "Using media impl: fsgst".
Unfortunately if these problems are caused by gstreamer's
audio playback, then use of gst-plugins-farsight won't help. :(
--
[EMAIL PROTECTED] (Kai Vehmanen)
Networking Technologies Laboratory, Nokia Research Center
________________________________
From: ext H. L. [mailto: [EMAIL PROTECTED]]
Sent: 31 May, 2006 19:22
To: Vehmanen Kai (Nokia-NRC/Helsinki)
Subject: Re: [Sofia-sip-devel] [gstreamer-rtp]
Hi Kai:
Thanks for your help!
Though the RTP trace is ok, I'm not able to hear the voice.
I think I downloaded gst-plugins-farsight package already. Can
you please tell me know to build sofcli using the fsgst?
thanks again!
On 5/30/06, [EMAIL PROTECTED] < [EMAIL PROTECTED]>
wrote:
Hi,
it's good to hear that it's working. The RTP port issue
is a known problem of basic gstreamer-RTP -> the RTP is
not
symmetric, which is really bad for NAT traversal. This
only
affects the basic "gstreamer" implementation. The
preferred
"fsgst" implementation however has symmetric RTP, but it
requires
installation of the the gst-plugins-farsight package, in
addition to gstreamer-0.10 .
As for test_media, can you see the packets on the
localhost
interface (tcpdump -i lo -n)? Also, which version of the
gst-plugins-base package?
________________________________
From: ext H. L. [mailto:[EMAIL PROTECTED]]
Sent: 31 May, 2006 00:45
To: Vehmanen Kai (Nokia-NRC/Helsinki)
Subject: Re: [Sofia-sip-devel] [gstreamer-rtp]
Thanks for the response! I loaded the
sofsip-cli-0.10, and I'm
able to make outgoing calls. Even I could see the two
sides exchange RTP
packets, I'm not able to hear the voice. One thing I
realized is that
the source port of RTP packet from sofsip-cli is not the
same as the
advertised port in SDP of the outgoing Invite, not sure
if this is
appropriate for the media application.
I tried to run test-media, but didn't have any
luck. I didn't
even see RTP packet. The following is the log:
[EMAIL PROTECTED] src]# SOFSIP_AUDIO=OSS
./test_media
** (test_media:10459): DEBUG: scheduler from=0
** (test_media:10459): DEBUG: scheduler to=2
** Message: Selecting media implementation:
Gstreamer-RTP
** (test_media:10459): DEBUG:
priv_verify_required_elements:191
** Message: Verifying GST element "mulawenc" ->
OK
** Message: Verifying GST element "mulawdec" ->
OK
** Message: Verifying GST element "dynudpsink" ->
OK
** Message: Verifying GST element "udpsrc" -> OK
** (test_media:10459): DEBUG:
ssc_media_class_init:124
** (test_media:10459): DEBUG:
ssc_media_gst_class_init:138
** (test_media:10459): DEBUG: ssc_media_init:167
** (test_media:10459): DEBUG: main:104
** (test_media:10459): DEBUG:
priv_static_capabilities_gst
** Message: Local SDP used for testing:
v=0
c=IN IP4 127.0.0.1
m=audio 5400 RTP/AVP 0
a=rtpmap:0 PCMU/8000
** Message: Remote SDP used for testing:
v=0
c=IN IP4 127.0.0.1
m=audio 5400 RTP/AVP 0
a=rtpmap:0 PCMU/8000
** (test_media:10459): DEBUG: priv_set_local_sdp
** (test_media:10459): DEBUG: priv_set_remote_sdp
** (test_media:10459): DEBUG: priv_activate_gst
** (test_media:10459): DEBUG: priv_activate_gst
Succesfully bound to local port 5400.
** (test_media:10459): DEBUG: priv_cb_ready
** (test_media:10459): DEBUG:
priv_update_rx_elements
** (test_media:10459): DEBUG:
priv_update_tx_elements
** Message: RTP destination is: 127.0.0.1:5400.
** Message: Starting the pipeline.
** Message: Phase 1 - running for 10secs...
** (test_media:10459): DEBUG: priv_deactivate_gst
** (test_media:10459): DEBUG: priv_activate_gst
** (test_media:10459): DEBUG: priv_activate_gst
** (test_media:10459): DEBUG:
ssc_media_gst_finalize
** (test_media:10459): DEBUG: scheduler from=2
** (test_media:10459): DEBUG: scheduler to=0
[EMAIL PROTECTED] src]# SOFSIP_AUDIO=OSS
./test_media
** (test_media:10486): DEBUG: scheduler from=0
** (test_media:10486): DEBUG: scheduler to=2
** Message: Selecting media implementation:
Gstreamer-RTP
** (test_media:10486): DEBUG:
priv_verify_required_elements:191
** Message: Verifying GST element "mulawenc" ->
OK
** Message: Verifying GST element "mulawdec" ->
OK
** Message: Verifying GST element "dynudpsink" ->
OK
** Message: Verifying GST element "udpsrc" -> OK
** (test_media:10486): DEBUG:
ssc_media_class_init:124
** (test_media:10486): DEBUG:
ssc_media_gst_class_init:138
** (test_media:10486): DEBUG: ssc_media_init:167
** (test_media:10486): DEBUG: main:104
** (test_media:10486): DEBUG:
priv_static_capabilities_gst
** Message: Local SDP used for testing:
v=0
c=IN IP4 127.0.0.1
m=audio 5400 RTP/AVP 0
a=rtpmap:0 PCMU/8000
** Message: Remote SDP used for testing:
v=0
c=IN IP4 127.0.0.1
m=audio 5400 RTP/AVP 0
a=rtpmap:0 PCMU/8000
** (test_media:10486): DEBUG: priv_set_local_sdp
** (test_media:10486): DEBUG: priv_set_remote_sdp
** (test_media:10486): DEBUG: priv_activate_gst
** (test_media:10486): DEBUG: priv_activate_gst
Succesfully bound to local port 5400.
** (test_media:10486): DEBUG: priv_cb_ready
** (test_media:10486): DEBUG:
priv_update_rx_elements
** (test_media:10486): DEBUG:
priv_update_tx_elements
** Message: RTP destination is: 127.0.0.1:5400.
** Message: Starting the pipeline.
** Message: Phase 1 - running for 10secs...
** (test_media:10486): DEBUG: priv_deactivate_gst
** (test_media:10486): DEBUG: priv_activate_gst
** (test_media:10486): DEBUG: priv_activate_gst
** (test_media:10486): DEBUG:
ssc_media_gst_finalize
** (test_media:10486): DEBUG: scheduler from=2
** (test_media:10486): DEBUG: scheduler to=0
On 5/30/06, [EMAIL PROTECTED]
< [EMAIL PROTECTED] >
wrote:
Hi,
On 27 May 2006, [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED] > wrote:
> The sofia-sip version i used is
the stable
1.11.8, sofsip_cli is
the snapshort you
> told me from the previous message(plain
gstreamer), as
well as the
gst_plugins_farsight.
> Gstreamer version on Fedora core 5 is
0.10.4_1. I
used the test_media
tool, and I did
> see RTP packets from tcpdump, but I
didn't hear
anything.
> The following is the gdb bt
trace.
thanks for the bug report. It seems there
was a stupid
bug in the
sofsip-cli
STUN code, which is now fixed (in the
sofsip-cli-0.9
release). Could you
try again, and let me know if it works
better?
If you still have problems, could you try
the following
(with a headset
attached, speak something to the
microphone :)):
cd sofsip-cli/src
SOFSIP_AUDIO=OSS ./test_media
SOFSIP_AUDIO=ALSA ./test_media
... on one of my devel machines, the
latter version does
not
work very well (I hear sound for a few
secs, but then it
stops). This
is probably some strange bug in
gstreamer's audio output
support. :(
--
[EMAIL PROTECTED] (Kai Vehmanen)
Networking Technologies Laboratory, Nokia
Research
Center
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