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 [334m 3837[00m 0x804e620 [31;01mERROR[00m [00m rtpbin jrtplib_c.cpp:77:int checkerror(int):[00m -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 __________________________________ Kennt man wirklich jeden über 3 Ecken? 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