Public feed #894444

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·published 2008-02-06 23:19 UTC
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voip@dl325:~/src/pymsn.devel$ python /home/voip/src/telepathy-python/telepathy-python/examples/call.py /home/voip/au.details david.laban@jabber.org.au
connecting
got handle 2 for david.laban@jabber.org.au
new streamed media channel
channel ready
requesting audio stream
MembersChanged (dbus.Array([], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.Array([dbus.UInt32(2L)], signature=dbus.Signature('u')), dbus.UInt32(0L), dbus.UInt32(0L))
MembersChanged (dbus.Array([], signature=dbus.Signature('u')), dbus.Array([dbus.UInt32(1L), dbus.UInt32(2L)], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.UInt32(1L), dbus.UInt32(8L))
channel closed
disconnecting



==============================================================================================

voip@dl325:~/src/stream-engine$ python /home/voip/src/telepathy-python/telepathy-python/examples/call.py /home/voip/d.l.au.details
connecting
new streamed media channel
channel ready
accepting incoming call
MembersChanged (dbus.Array([dbus.UInt32(1L)], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.UInt32(0L), dbus.UInt32(0L))
MembersChanged (dbus.Array([], signature=dbus.Signature('u')), dbus.Array([dbus.UInt32(1L), dbus.UInt32(2L)], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.Array([], signature=dbus.Signature('u')), dbus.UInt32(2L), dbus.UInt32(0L))
channel closed
disconnecting


===========================================================================================


voip@dl325:~/src/pymsn.devel$ ~/libexec/telepathy-gabble || killall telepathy-gabble && ~/libexec/telepathy-gabble
(telepathy-gabble:22388): tp-glib-DEBUG: started version 0.7.2.1 (telepathy-glib version 0.7.1.1)
** (telepathy-gabble:22388): DEBUG: connection_iq_disco_cb: caps now 895
** (telepathy-gabble:22388): DEBUG: connection_iq_disco_cb: caps now 869
(telepathy-gabble:22388): tp-glib-DEBUG: no connections, and timed out
(telepathy-gabble:22958): tp-glib-DEBUG: started version 0.7.2.1 (telepathy-glib version 0.7.1.1)
(telepathy-gabble:22958): tp-glib-DEBUG: no connections, and timed out






===================================================================================================






voip@dl325:~/src/pymsn.devel$ /usr/bin/telepathy-stream-engine
** (telepathy-stream-engine:23003): DEBUG: not using realtime scheduling, enable through STREAM_ENGINE_REALTIME env
** (telepathy-stream-engine:23003): DEBUG: Requesting org.freedesktop.Telepathy.StreamEngine
** (telepathy-stream-engine:23003): DEBUG: registering StreamEngine at /org/freedesktop/Telepathy/StreamEngine
** (telepathy-stream-engine:23003): DEBUG: started
** (telepathy-stream-engine:23003): DEBUG: HandleChannel called
** (telepathy-stream-engine:23003): DEBUG: GetSessionHandlers returned 0 sessions
** (telepathy-stream-engine:23003): DEBUG: adding session handler /org/freedesktop/Telepathy/Connection/gabble/jabber/alsuren_40jabber_2eorg_2eau_2fTelepathy/MediaChannel0/MediaSession2, type rtp
farsight-Message: looking for plugins in /usr/lib/farsight-0.1-2
opening module /usr/lib/farsight-0.1-2/librtp-session.so: succeeded
** (telepathy-stream-engine:23003): DEBUG: plugin details:
 name: rtp-session
 description: Farsight RTP plugin
 author: Farsight Project

** (telepathy-stream-engine:23003): DEBUG: Calling MediaSessionHandler::Ready -->
** (telepathy-stream-engine:23003): DEBUG: <-- Returned from MediaSessionHandler::Ready
** (telepathy-stream-engine:23003): DEBUG: got nat-traversal = gtalk-p2p
** (telepathy-stream-engine:23003): DEBUG: Adding stream, media_type=0, direction=3
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) make_src: making audio src with alsasrc element
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) make_sink: making audio sink with alsasink element
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) tp_stream_engine_stream_go: setting source on Farsight stream
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: ensure_local_codecs: media type is 0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: looking for /home/voip/.farsight/gstelements.conf
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: looking for /etc/farsight/gstelements.conf
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: skipping rtpdepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: skipping rtpdec
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: skipping rtspsrc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: skipping rtpdtmfsrc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec GSM with pt 3, send_pipeline 0x815fa20, receive_pipeline 0x815fab0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80d0800:5:gsmenc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x808eec0:5:rtpgsmpay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80d0890:5:gsmdec
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x808ee30:5:rtpgsmdepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec PCMA with pt 8, send_pipeline 0x8163470, receive_pipeline 0x8163250
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8082480:5:alawenc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8091a40:5:rtppcmapay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8082510:5:alawdec
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8091890:5:rtppcmadepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec PCMU with pt 0, send_pipeline 0x8163220, receive_pipeline 0x8169440
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x808e920:5:mulawenc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80919b0:5:rtppcmupay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x808e9b0:5:mulawdec
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8091920:5:rtppcmudepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec MPEG4-GENERIC with pt -1, send_pipeline 0x8169660, receive_pipeline 0x8162500
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80f61b0:4:faac
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8094c80:6:rtpmp4gpay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80f6120:7:faad
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8094bf0:6:rtpmp4gdepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec MPEG4-GENERIC with pt -1, send_pipeline 0x8162100, receive_pipeline 0x8162b00
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80fea40:4:ffenc_mpeg4
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8094c80:7:rtpmp4gpay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8108630:4:ffdec_mpeg4
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8094bf0:7:rtpmp4gdepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec SPEEX with pt -1, send_pipeline 0x8163110, receive_pipeline 0x8162c30
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80a0510:4:speexenc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8094d10:4:rtpspeexpay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80a05a0:4:speexdec
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8094da0:4:rtpspeexdepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: adding codec VORBIS with pt -1, send_pipeline 0x815f3a0, receive_pipeline 0x815f7d0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80ca000:4:vorbisenc
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8099090:4:rtpvorbispay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: pipeline:
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x80ca120:4:vorbisparse
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: 0x8099000:4:rtpvorbisdepay
(telepathy-stream-engine:23003): farsight-rtp-DEBUG:

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: skipping codec audio/MSGSM, no encoding name specified (pt: -1 clock_rate:0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Trying to load /home/voip/.farsight/gstcodecs.conf
(telepathy-stream-engine:23003): farsight-DEBUG: Unable to read file /home/voip/.farsight/gstcodecs.conf: No such file or directory
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Trying to load /etc/farsight/gstcodecs.conf
(telepathy-stream-engine:23003): farsight-DEBUG: Unable to read file /etc/farsight/gstcodecs.conf: No such file or directory
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: ensure_local_codecs (511): could not load codecs configuration file
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_set_source (1737): setting src
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) tp_stream_engine_stream_go: setting sink on Farsight stream
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink (1847): setting sink alsasink0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink: No codec bin present, setting new sink for future use
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_stream_properties: setting farsight transmitter to libjingle
farsight-Message: looking for plugins in /usr/lib/farsight-0.1-2
opening module /usr/lib/farsight-0.1-2/libjingle-transmitter: succeeded
farsight-rtp-Message: Preparing transmitter
farsight-transmitter-Message: Init and running jinglep2p to prepare native candiates, using stun stun.l.google.com : 19302
farsight-transmitter-Message: No socket created yet! SocketState not connected
farsight-rtp-Message: connect state changed to 1
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) cb_fs_state_changed: stream 0x8151028, state: connecting, direction: none
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_build_base_pipeline (2066): creating core RTP pipeline
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: added transmitter_src 0x819d028 to pipeline 0x81a0040
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec GSM [3]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec PCMA [8]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec PCMU [0]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec telephone-event [96]
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) prepare_transports: calling MediaStreamHandler::Ready
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_stream_playing: 0
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_stream_sending: 0
farsight-rtp-Message: send codec bin not created yet, will set sending later
OnRequestsignaling called, sending SignalingReady
** (telepathy-stream-engine:23003): DEBUG: HandleChannel called
** (telepathy-stream-engine:23003): DEBUG: GetSessionHandlers replied:
** (telepathy-stream-engine:23003): DEBUG:   - session /org/freedesktop/Telepathy/Connection/gabble/jabber/david_2elaban_40jabber_2eorg_2eau_2fTelepathy/MediaChannel0/MediaSession2
** (telepathy-stream-engine:23003): DEBUG:     type rtp
** (telepathy-stream-engine:23003): DEBUG: adding session handler /org/freedesktop/Telepathy/Connection/gabble/jabber/david_2elaban_40jabber_2eorg_2eau_2fTelepathy/MediaChannel0/MediaSession2, type rtp
** (telepathy-stream-engine:23003): DEBUG: plugin details:
 name: rtp-session
 description: Farsight RTP plugin
 author: Farsight Project

** (telepathy-stream-engine:23003): DEBUG: Calling MediaSessionHandler::Ready -->
** (telepathy-stream-engine:23003): DEBUG: <-- Returned from MediaSessionHandler::Ready
** (telepathy-stream-engine:23003): DEBUG: got nat-traversal = gtalk-p2p
** (telepathy-stream-engine:23003): DEBUG: Adding stream, media_type=0, direction=3
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) make_src: making audio src with alsasrc element
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) make_sink: making audio sink with alsasink element
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) tp_stream_engine_stream_go: setting source on Farsight stream
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: ensure_local_codecs: media type is 0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Trying to load /home/voip/.farsight/gstcodecs.conf
(telepathy-stream-engine:23003): farsight-DEBUG: Unable to read file /home/voip/.farsight/gstcodecs.conf: No such file or directory
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Trying to load /etc/farsight/gstcodecs.conf
(telepathy-stream-engine:23003): farsight-DEBUG: Unable to read file /etc/farsight/gstcodecs.conf: No such file or directory
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: ensure_local_codecs (511): could not load codecs configuration file
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_set_source (1737): setting src
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) tp_stream_engine_stream_go: setting sink on Farsight stream
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink (1847): setting sink alsasink1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink: No codec bin present, setting new sink for future use
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_stream_properties: setting farsight transmitter to libjingle
farsight-rtp-Message: Preparing transmitter
farsight-transmitter-Message: Init and running jinglep2p to prepare native candiates, using stun stun.l.google.com : 19302
farsight-transmitter-Message: No socket created yet! SocketState not connected
farsight-rtp-Message: connect state changed to 1
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) cb_fs_state_changed: stream 0x8151100, state: connecting, direction: none
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: farsight_rtp_stream_build_base_pipeline (2066): creating core RTP pipeline
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: added transmitter_src 0x819d200 to pipeline 0x81a0130
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec GSM [3]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec PCMA [8]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec PCMU [0]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec telephone-event [96]
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) prepare_transports: calling MediaStreamHandler::Ready
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_remote_codecs: called
** Message: set_remote_codecs: adding remote codec GSM [3]
** Message: set_remote_codecs: adding remote codec PCMA [8]
** Message: set_remote_codecs: adding remote codec PCMU [0]
** Message: set_remote_codecs: adding remote codec telephone-event [96]
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: remote_codec GSM 8000
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: remote_codec PCMA 8000
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: remote_codec PCMU 8000
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: remote_codec telephone-event 8000
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Remote codec 3: audio GSM clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Have local codec in the same PT, lets try it first
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Using default codec negotiation function
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Negotiated codec 3: audio GSM clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Remote codec 8: audio PCMA clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Have local codec in the same PT, lets try it first
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Using default codec negotiation function
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Negotiated codec 8: audio PCMA clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Remote codec 0: audio PCMU clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Have local codec in the same PT, lets try it first
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Using default codec negotiation function
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Negotiated codec 0: audio PCMU clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Remote codec 96: audio telephone-event clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Encoding names dont match, local: PCMU, remote: telephone-event
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Encoding names dont match, local: GSM, remote: telephone-event
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Encoding names dont match, local: PCMA, remote: telephone-event
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Could not find a valid intersection... for codec 96: audio telephone-event clock:8000 channels:1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: caps are: application/x-rtp, encoding-name=(string)PCMU, clock-rate=(int)8000, media=(string)audio, payload=(int)0, channels=(int)1

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: caps are: application/x-rtp, encoding-name=(string)GSM, clock-rate=(int)8000, media=(string)audio, payload=(int)3, channels=(int)1

(telepathy-stream-engine:23003): farsight-rtp-DEBUG: caps are: application/x-rtp, encoding-name=(string)PCMA, clock-rate=(int)8000, media=(string)audio, payload=(int)8, channels=(int)1

farsight-rtp-Message: farsight_rtp_stream_build_send_pipeline (2376): creating send pipeline with codec 3
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: creating send codec bin for id 3, pipeline_factory 0x815fa20
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: checking if gsmenc is in config list
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: checking if rtpgsmpay is in config list
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: linking gsmenc0 and rtpgsmpay0 with caps audio/x-gsm, rate=(int)8000, channels=(int)1
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: linking src 0x815a220 to codec bin 0x81bb000 with caps NULL
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Found audio/telephone-event with PT 96

** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec GSM [3]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec PCMA [8]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec PCMU [0]
** (telepathy-stream-engine:23003): DEBUG: fs_codecs_to_tp: adding codec telephone-event [96]
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_stream_playing: 0
** (telepathy-stream-engine:23003): DEBUG: stream 1 (audio) set_stream_sending: 0
(telepathy-stream-engine:23003): farsight-rtp-DEBUG: Setting sending on 0
OnRequestsignaling called, sending SignalingReady
OnRequestsignaling called, sending SignalingReady
OnRequestsignaling called, sending SignalingReady
Segmentation fault (core dumped)