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

Tom Huntter wrote:
> Hello,
> 
>  I'm new to farsight. To understand the basics I
> compiled the files in the farsight/tests/ directory,
> but when I for example run test-rtp-3, wireshark
> doesn't show any network activity.
> Could you please help me and tell me what I'm missing?
> 
> I use CVS/darcs versions of farsight and gstreamer.
> 
> Thanks,
> Thomas
> 
> This is the output of ./test-rtp-3 10.0.0.1 3000
> 
> (lt-test-rtp-3:3851): GStreamer-WARNING **: Failed to
> load plugin
> '/usr/local/lib/gstreamer-0.10/libgstgconfelements2.so':
> /usr/local/lib/gstreamer-0.10/libgstgconfelements2.so:
> undefined symbol: gst_video_orientation_set_hflip
> 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-3:3837): farsight-rtp-DEBUG: looking for
> /root/.farsight/gstelements.conf
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: looking for
> /usr/local/etc/farsight/gstelements.conf
> 
> (lt-test-rtp-3:3837): farsight-rtp-WARNING **: Not
> enough information in rtp caps
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: adding codec
> PCMA with pt 8, send_pipeline 0x80cc470,
> receive_pipeline 0x80b7370
> pipeline: 0x80a2ad0:4:alawenc 0x8086ad0:4:rtppcmapay 
> pipeline: 0x80a2b60:4:alawdec 0x8086920:4:rtppcmadepay
> 
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: adding codec
> PCMU with pt 0, send_pipeline 0x8060f20,
> receive_pipeline 0x8061010
> pipeline: 0x80af630:4:mulawenc 0x8086a40:4:rtppcmupay 
> pipeline: 0x80af6c0:4:mulawdec
> 0x80869b0:4:rtppcmudepay 
> 
> (lt-test-rtp-3:3837): farsight-rtp-WARNING **: Not
> enough information in rtp caps
> 
> (lt-test-rtp-3:3837): farsight-rtp-WARNING **: Not
> enough information in rtp caps
> 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-3:3837): 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
> 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-3:3837): farsight-transmitter-DEBUG: The
> socket was created
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG:
> could not bind port 7078
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG: The
> socket was created
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG:
> could not bind port 7080
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG: The
> socket was created
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG:
> could not bind port 7082
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG: The
> socket was created
> (lt-test-rtp-3:3837): farsight-transmitter-DEBUG:
> bound to port 7084
> (lt-test-rtp-3:3837): farsight-DEBUG: Interface:  eth0
> (lt-test-rtp-3:3837): farsight-DEBUG: IP Address:
> 10.0.0.46
> (lt-test-rtp-3:3837): farsight-DEBUG: Interface:  lo
> (lt-test-rtp-3:3837): farsight-DEBUG: IP Address:
> 127.0.0.1
> (lt-test-rtp-3:3837): farsight-DEBUG: Ignoring
> loopback interface
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: Called
> farsight_rtp_stream_new_native_candidate
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: Native
> candidates found, adding to list
> native_candidates_prepared: preparation-complete:
> stream=0x8060000
> ** Message: Local transport candidate: L1 1 UDP RTP
> 10.0.0.46:7084, pref 1.000000
> farsight-rtp-Message: connect state changed to 2
> ** Message: state_changed: 0x8060000 connected
> 
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG:
> farsight_rtp_stream_build_base_pipeline (1851):
> creating core RTP pipeline
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: added
> transmitter_src 0x80d0020 to pipeline 0x80d4038
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG:
> farsight_rtp_stream_set_active_codec (1042): called to
> change codec from -1 to 8
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: remote_codec
> PCMU 8000
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: remote_codec
> PCMA 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-3:3837): farsight-rtp-DEBUG: We are now
> PLAYING
> 
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: Set_state
> result was 1
> 
> 0x810b3b0
> new_active_candidate_pair: new-native-candidate-pair:
> stream=0x8060000
> 
> (lt-test-rtp-3:3837): GLib-GObject-WARNING **:
> IA__g_object_set_valist: object class
> `GstAudioTestSrc' has no property named `latency-time'
> (lt-test-rtp-3:3837): 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
> 0x8114c18
> 0:00:00.214497000  3837 0x804e620
> ERROR               rtpbin
> jrtplib_c.cpp:77:int checkerror(int): -4 Element
> already exists in hash table
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: creating
> send codec bin for id 8, pipeline_factory 0x80cc470
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: checking if
> alawenc is in config list
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: checking if
> rtppcmapay is in config list
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: linking
> alawenc0 and rtppmcapay0 with caps audio/x-alaw,
> channels=(int)1, rate=(int)8000
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: linking src
> 0x810e210 to codec bin 0x80600b8 with caps NULL
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG: Remote end
> does not have audio/telephone-event
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG:
> farsight_rtp_stream_set_sink (1633): setting sink
> alsasink
> (lt-test-rtp-3:3837): farsight-rtp-DEBUG:
> farsight_rtp_stream_set_sink: No codec bin present,
> setting new sink for future use
> 
> 
> 
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