Hi everybody , thank's for the advice and specially to Mr .olivier . i've
compiled the example test-rtp-3 , and finnaly i'm capturing packets with
Tcpdump . but nothing happens anymore , i mean that the test stop's in a
strange manner . i'm sending you the log of the execution hopping to receive an
ansewer about how to use this code to send or receive a stream .
//
./test-rtp-3 192.168.1.114 8548
farsight-Message: looking for plugins in
/home/g545952/Desktop/fado//lib/farsight-0.1-2
opening module
/home/g545952/Desktop/fado//lib/farsight-0.1-2/librtp-session.so: succeeded
protocol details:
name: rtp-session
description: Farsight RTP plugin
author: Farsight Project
farsight-Message: looking for plugins in
/home/g545952/Desktop/fado//lib/farsight-0.1-2
opening module
/home/g545952/Desktop/fado//lib/farsight-0.1-2/librawudp-transmitter.so:
succeeded
(lt-test-rtp-3:6724): GLib-GObject-WARNING **: IA__g_object_set_valist: object
class `FarsightRTPStream' has no property named `stun_ip'
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: ensure_local_codecs: media type is 0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: looking for
/home/g545952/.farsight/gstelements.conf
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: looking for
/home/g545952/Desktop/fado//etc/farsight/gstelements.conf
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtpdepay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtpdec
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtspsrc
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtpdtmfsrc
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec VORBIS with pt -1,
send_pipeline 0x810ad00, receive_pipeline 0x810ac60
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8082630:4:vorbisenc
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093ec0:4:rtpvorbispay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80825a0:4:vorbisdec
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093f50:4:rtpvorbisdepay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec MPEG4-GENERIC with pt
-1, send_pipeline 0x810a8c0, receive_pipeline 0x810b270
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80f15a0:4:ffenc_mpeg4
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809cad0:4:rtpmp4gpay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80e81b0:4:ffdec_mpeg4
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809cb60:4:rtpmp4gdepay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping codec audio/unknown, no
encoding name specified (pt: 14 clock_rate:90000
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec MPA with pt -1,
send_pipeline 0x810b6d0, receive_pipeline 0x810a700
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80f5ad0:6:ffenc_mp2
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f3f0:6:rtpmpapay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8088920:6:mp3parse
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f480:6:rtpmpadepay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec PCMU with pt 0,
send_pipeline 0x80e0bd0, receive_pipeline 0x810a650
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093a40:5:mulawenc
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f5a0:5:rtppcmupay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80939b0:5:mulawdec
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f630:5:rtppcmudepay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec PCMA with pt 8,
send_pipeline 0x810b4b0, receive_pipeline 0x812f540
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093920:5:alawenc
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f510:5:rtppcmapay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093890:5:alawdec
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f6c0:5:rtppcmadepay
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Trying to load
/home/g545952/.farsight/gstcodecs.conf
(lt-test-rtp-3:6724): farsight-DEBUG: Unable to read file
/home/g545952/.farsight/gstcodecs.conf: Aucun fichier ou répertoire de ce type
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Trying to load
/home/g545952/Desktop/fado//etc/farsight/gstcodecs.conf
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio speex
clock:8000 channels:0 could not be matched with an installed
encoder/decoder/payloader/depayloader quatuor
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio iLBC
clock:8000 channels:0 mode=30 could not be matched with an installed
encoder/decoder/payloader/depayloader quatuor
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio AMR
clock:8000 channels:0 could not be matched with an installed
encoder/decoder/payloader/depayloader quatuor
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio GSM
clock:8000 channels:0 could not be matched with an installed
encoder/decoder/payloader/depayloader quatuor
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio G729
clock:8000 channels:0 could not be matched with an installed
encoder/decoder/payloader/depayloader quatuor
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio PCMU
clock:16000 channels:0 could not be matched with an installed
encoder/decoder/payloader/depayloader quatuor
** Message: codec: 8: PCMA/8000 found
** Message: codec: 96: MPA/90000 found
** Message: codec: 0: PCMU/8000 found
** Message: codec: 97: telephone-event/8000 found
farsight-rtp-Message: Preparing transmitter
farsight-rtp-Message: connect state changed to 1
** Message: state_changed: 0x8063000 connecting
farsight-transmitter-Message: Media type is 0
(lt-test-rtp-3:6724): farsight-transmitter-DEBUG: The socket was created
(lt-test-rtp-3:6724): farsight-transmitter-DEBUG: bound to port 7078
(lt-test-rtp-3:6724): farsight-DEBUG: Interface: lo
(lt-test-rtp-3:6724): farsight-DEBUG: IP Address: 127.0.0.1
(lt-test-rtp-3:6724): farsight-DEBUG: Ignoring loopback interface
(lt-test-rtp-3:6724): farsight-DEBUG: Interface: eth1
(lt-test-rtp-3:6724): farsight-DEBUG: IP Address: 192.168.1.9
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Called
farsight_rtp_stream_new_native_candidate
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Native candidates found, adding to
list
native_candidates_prepared: preparation-complete: stream=0x8063000
** Message: Local transport candidate: L1 1 UDP RTP 192.168.1.9:7078, pref
1,000000
farsight-rtp-Message: connect state changed to 2
** Message: state_changed: 0x8063000 connected
(lt-test-rtp-3:6724): farsight-rtp-DEBUG:
farsight_rtp_stream_build_base_pipeline (2067): creating core RTP pipeline
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: added transmitter_src 0x813a048 to
pipeline 0x813e060
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: farsight_rtp_stream_set_active_codec
(1193): called to change codec from -1 to 8
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec PCMU 8000
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec PCMA 8000
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec MPA 90000
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec telephone-event 8000
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 0: audio PCMU clock:8000
channels:0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Have local codec in the same PT, lets
try it first
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Using default codec negotiation
function
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Negotiated codec 0: audio PCMU
clock:8000 channels:0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 8: audio PCMA clock:8000
channels:0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Have local codec in the same PT, lets
try it first
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Using default codec negotiation
function
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Negotiated codec 8: audio PCMA
clock:8000 channels:0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 96: audio MPA
clock:90000 channels:0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Have local codec in the same PT, lets
try it first
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Using default codec negotiation
function
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Negotiated codec 96: audio MPA
clock:90000 channels:0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 97: audio
telephone-event clock:8000 channels:1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Encoding names dont match, local:
PCMU, remote: telephone-event
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Encoding names dont match, local:
PCMA, remote: telephone-event
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Encoding names dont match, local:
MPA, remote: telephone-event
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Could not find a valid
intersection... for codec 97: audio telephone-event clock:8000 channels:1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: caps are: application/x-rtp,
encoding-name=(string)PCMU, clock-rate=(int)8000, media=(string)audio,
payload=(int)0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: caps are: application/x-rtp,
encoding-name=(string)PCMA, clock-rate=(int)8000, media=(string)audio,
payload=(int)8
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: caps are: application/x-rtp,
encoding-name=(string)MPA, clock-rate=(int)90000, media=(string)audio,
payload=(int)96
farsight-rtp-Message: farsight_rtp_stream_build_send_pipeline (2662): creating
send pipeline with codec 8
farsight-rtp-Message: No source has been set yet, send pipeline build for later
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: We are now PLAYING
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Set_state result was 1
0x8147400
new_active_candidate_pair: new-native-candidate-pair: stream=0x8063000
** (lt-test-rtp-3:6724): DEBUG: Sink needs buffer, must wait for buffering to
stop
(lt-test-rtp-3:6724): GLib-GObject-WARNING **: IA__g_object_set_valist: object
class `GstAudioTestSrc' has no property named `latency-time'
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: farsight_rtp_stream_set_source
(1738): setting src
farsight-rtp-Message: farsight_rtp_stream_build_send_pipeline (2662): creating
send pipeline with codec 8
0x8150520
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: creating send codec bin for id 8,
pipeline_factory 0x810b4b0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: checking if alawenc is in config list
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: checking if rtppcmapay is in config
list
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: linking alawenc0 and rtppmcapay0 with
caps audio/x-alaw, channels=(int)1, rate=(int)8000
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: linking src 0x8148d80 to codec bin
0x8155800 with caps NULL
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found audio/telephone-event with PT 97
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Looking for PCMA or PCMU in the
remote codecs
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMU codec with PT 0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMA codec with PT 8
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found remote_pcma = 1 - remote_pcmu =
1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found both the dtmfsrc and rtpdtmfmux
installed
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMA found locally? 1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMU found locally? 1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Looking for PCMA or PCMU in the
remote codecs
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMU codec with PT 0
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMA codec with PT 8
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found remote_pcma = 1 - remote_pcmu =
1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found both the dtmfsrc and rtpdtmfmux
installed
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMA found locally? 1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMU found locally? 1
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink (1848):
setting sink alsasink
(lt-test-rtp-3:6724): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink: No
codec bin present, setting new sink for future use
//
but the problemeis that noting happens after this . can someone please guide me
how to continue ( i mean how to send or receive the stream )
regards
Souilem Fadi
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