Hi,
I tried now two combinations:
1. all latest stable
2. farsight darcs, gstreamer and gst-plugins-farsight
latest stable
Still no packets showing on wireshark..
Thomas
Output for test-rtp-3 scenario 1:
farsight-Message: looking for plugins in
/usr/local/lib/farsight-0.1-2
opening module
/usr/local/lib/farsight-0.1-2/librtp-session.so:
succeeded
protocol details:
name: rtp-session
description: Farsight RTP plugin
author: Farsight Project
farsight-rtp-Message: Media type is 0
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: looking for
/root/.farsight/gstelements.conf
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: looking for
/usr/local/etc/farsight/gstelements.conf
(lt-test-rtp-3:19722): GStreamer-CRITICAL **:
gst_caps_unref: assertion `caps != NULL' failed
(lt-test-rtp-3:19722): farsight-rtp-WARNING **: Not
enough information in rtp caps
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: adding
codec speex with pt 97, send_pipeline 0x8061360,
receive_pipeline 0x8061340
pipeline: 0x807af50:4:speexenc 0x8071000:4:rtpspeexpay
pipeline: 0x807aec0:4:speexdec
0x806c750:4:rtpspeexdepay
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: adding
codec PCMU with pt 0, send_pipeline 0x80c5490,
receive_pipeline 0x8061590
pipeline: 0x80a4ad0:4:mulawenc 0x8075800:4:rtppcmupay
pipeline: 0x80a4a40:4:mulawdec
0x8075890:4:rtppcmudepay
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: adding
codec PCMA with pt 8, send_pipeline 0x80bd800,
receive_pipeline 0x8061380
pipeline: 0x808b1b0:4:alawenc 0x8071750:4:rtppcmapay
pipeline: 0x808b120:4:alawdec 0x8075920:4:rtppcmadepay
farsight-Message: looking for plugins in
/usr/local/lib/farsight-0.1-2
opening module
/usr/local/lib/farsight-0.1-2/librawudp-transmitter.so:
succeeded
(lt-test-rtp-3:19722): GLib-GObject-WARNING **:
IA__g_object_set_valist: object class
`FarsightRTPStream' has no property named `stun_ip'
** Message: codec: 8: PCMA/8000 found
** Message: codec: 0: PCMU/8000 found
** Message: codec: 97: speex/8000 found
farsight-rtp-Message: Preparing transmitter
farsight-rtp-Message: connect state changed to 1
** Message: state_changed: 0x8060000 connecting
farsight-transmitter-Message: Media type is 0
(lt-test-rtp-3:19722): farsight-transmitter-DEBUG: The
socket was created
(lt-test-rtp-3:19722): farsight-transmitter-DEBUG:
bound to port 7078
(lt-test-rtp-3:19722): farsight-DEBUG: Interface:
eth0
(lt-test-rtp-3:19722): farsight-DEBUG: IP Address:
10.147.67.83
(lt-test-rtp-3:19722): farsight-DEBUG: Interface: lo
(lt-test-rtp-3:19722): farsight-DEBUG: IP Address:
127.0.0.1
(lt-test-rtp-3:19722): farsight-DEBUG: Ignoring
loopback interface
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: Called
farsight_rtp_stream_new_native_candidate
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: Native
candidates found, adding to list
native_candidates_prepared: preparation-complete:
stream=0x8060000
** Message: Local transport candidate: L1 1 UDP RTP
10.147.67.83:7078, pref 1.000000
farsight-rtp-Message: connect state changed to 2
** Message: state_changed: 0x8060000 connected
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
farsight_rtp_stream_build_base_pipeline (1762):
creating core RTP pipeline
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: added
transmitter_src 0x80c70c8 to pipeline 0x80ca080
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
farsight_rtp_stream_set_active_codec (1010): called to
change codec from -1 to 8
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
farsight_rtp_stream_set_active_codec: this does not
work yet, returning
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
remote_codec PCMA 8000
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
remote_codec PCMU 8000
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
remote_codec speex 8000
farsight-rtp-Message:
farsight_rtp_stream_build_send_pipeline (2039):
creating send pipeline with codec 8
farsight-rtp-Message: No source has been set yet, send
pipeline build for later
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: We are now
PLAYING
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: Set_state
result was 1
0x8102ae0
new_active_candidate_pair: new-native-candidate-pair:
stream=0x8060000
(lt-test-rtp-3:19722): GLib-GObject-WARNING **:
IA__g_object_set_valist: object class
`GstAudioTestSrc' has no property named `latency-time'
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
farsight_rtp_stream_set_source (1479): setting src
farsight-rtp-Message:
farsight_rtp_stream_build_send_pipeline (2039):
creating send pipeline with codec 8
0x810dbb0
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: creating
send codec bin for id 8, pipeline_factory 0x80bd800
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: checking if
alawenc is in config list
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: checking if
rtppcmapay is in config list
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: linking
alawenc0 and rtppmcapay0 with caps audio/x-alaw,
channels=(int)1, rate=(int)8000
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: linking src
0x81073b8 to codec bin 0x8110800 with caps NULL
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: Remote end
does not have audio/telephone-event
(lt-test-rtp-3:19722): farsight-rtp-DEBUG: Not
building DTMF pipeline
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
farsight_rtp_stream_set_sink (1586): setting sink
alsasink
(lt-test-rtp-3:19722): farsight-rtp-DEBUG:
farsight_rtp_stream_set_sink: No codec bin present,
setting new sink for future use
Output for test-rtp-3 scenario 2:
farsight-Message: looking for plugins in
/usr/local/lib/farsight-0.1-2
opening module
/usr/local/lib/farsight-0.1-2/librtp-session.so:
succeeded
protocol details:
name: rtp-session
description: Farsight RTP plugin
author: Farsight Project
farsight-rtp-Message: Media type is 0
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: looking for
/root/.farsight/gstelements.conf
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: looking for
/usr/local/etc/farsight/gstelements.conf
(lt-test-rtp-3:9885): farsight-rtp-WARNING **: Not
enough information in rtp caps
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: adding codec
speex with pt 97, send_pipeline 0x8061360,
receive_pipeline 0x8061340
pipeline: 0x807af50:4:speexenc 0x8071000:4:rtpspeexpay
pipeline: 0x807aec0:4:speexdec
0x806c750:4:rtpspeexdepay
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: adding codec
PCMU with pt 0, send_pipeline 0x80c5490,
receive_pipeline 0x8061590
pipeline: 0x80a4ad0:4:mulawenc 0x8075800:4:rtppcmupay
pipeline: 0x80a4a40:4:mulawdec
0x8075890:4:rtppcmudepay
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: adding codec
PCMA with pt 8, send_pipeline 0x80bd800,
receive_pipeline 0x8061380
pipeline: 0x808b1b0:4:alawenc 0x8071750:4:rtppcmapay
pipeline: 0x808b120:4:alawdec 0x8075920:4:rtppcmadepay
farsight-Message: looking for plugins in
/usr/local/lib/farsight-0.1-2
opening module
/usr/local/lib/farsight-0.1-2/librawudp-transmitter.so:
succeeded
(lt-test-rtp-3:9885): GLib-GObject-WARNING **:
IA__g_object_set_valist: object class
`FarsightRTPStream' has no property named `stun_ip'
** Message: codec: 8: PCMA/8000 found
** Message: codec: 0: PCMU/8000 found
** Message: codec: 97: speex/8000 found
farsight-rtp-Message: Preparing transmitter
farsight-rtp-Message: connect state changed to 1
** Message: state_changed: 0x8060000 connecting
farsight-transmitter-Message: Media type is 0
(lt-test-rtp-3:9885): farsight-transmitter-DEBUG: The
socket was created
(lt-test-rtp-3:9885): farsight-transmitter-DEBUG:
bound to port 7078
(lt-test-rtp-3:9885): farsight-DEBUG: Interface: eth0
(lt-test-rtp-3:9885): farsight-DEBUG: IP Address:
10.147.67.83
(lt-test-rtp-3:9885): farsight-DEBUG: Interface: lo
(lt-test-rtp-3:9885): farsight-DEBUG: IP Address:
127.0.0.1
(lt-test-rtp-3:9885): farsight-DEBUG: Ignoring
loopback interface
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: Called
farsight_rtp_stream_new_native_candidate
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: Native
candidates found, adding to list
native_candidates_prepared: preparation-complete:
stream=0x8060000
** Message: Local transport candidate: L1 1 UDP RTP
10.147.67.83:7078, pref 1.000000
farsight-rtp-Message: connect state changed to 2
** Message: state_changed: 0x8060000 connected
(lt-test-rtp-3:9885): farsight-rtp-DEBUG:
farsight_rtp_stream_build_base_pipeline (1851):
creating core RTP pipeline
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: added
transmitter_src 0x80c70c0 to pipeline 0x80ca078
(lt-test-rtp-3:9885): farsight-rtp-DEBUG:
farsight_rtp_stream_set_active_codec (1042): called to
change codec from -1 to 8
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: remote_codec
PCMU 8000
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: remote_codec
PCMA 8000
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: remote_codec
speex 8000
farsight-rtp-Message:
farsight_rtp_stream_build_send_pipeline (2129):
creating send pipeline with codec 8
farsight-rtp-Message: No source has been set yet, send
pipeline build for later
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: We are now
PLAYING
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: Set_state
result was 1
0x8102b48
new_active_candidate_pair: new-native-candidate-pair:
stream=0x8060000
(lt-test-rtp-3:9885): GLib-GObject-WARNING **:
IA__g_object_set_valist: object class
`GstAudioTestSrc' has no property named `latency-time'
(lt-test-rtp-3:9885): farsight-rtp-DEBUG:
farsight_rtp_stream_set_source (1525): setting src
farsight-rtp-Message:
farsight_rtp_stream_build_send_pipeline (2129):
creating send pipeline with codec 8
0x810dc10
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: creating
send codec bin for id 8, pipeline_factory 0x80bd800
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: checking if
alawenc is in config list
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: checking if
rtppcmapay is in config list
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: linking
alawenc0 and rtppmcapay0 with caps audio/x-alaw,
channels=(int)1, rate=(int)8000
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: linking src
0x81073a8 to codec bin 0x80600b8 with caps NULL
(lt-test-rtp-3:9885): farsight-rtp-DEBUG: Remote end
does not have audio/telephone-event
(lt-test-rtp-3:9885): farsight-rtp-DEBUG:
farsight_rtp_stream_set_sink (1633): setting sink
alsasink
(lt-test-rtp-3:9885): farsight-rtp-DEBUG:
farsight_rtp_stream_set_sink: No codec bin present,
setting new sink for future use
--- Philippe Kalaf <[EMAIL PROTECTED]>
wrote:
> Hi,
>
> You might have trouble getting things to run with
> GStreamer CVS and that
> one contains some relatively important state change
> changes. Use
> Gstreamer 10.12, and you might want to update your
> gst-plugins-farsight
> as well.
>
> BR,
> Philippe
___________________________________________________________
Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de
-------------------------------------------------------------------------
This SF.net email is sponsored by DB2 Express
Download DB2 Express C - the FREE version of DB2 express and take
control of your XML. No limits. Just data. Click to get it now.
http://sourceforge.net/powerbar/db2/
_______________________________________________
Farsight-devel mailing list
Farsight-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/farsight-devel
