Hi, I got a weird issue when I dialed an extension and listen to a recorded voice mail greeting message.
After playing a couple of time of the greeting, the FS printed the warning of "sample rate not matching", then
send the audio to a different remote RTP port. See the log below,
2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@16000hz 1 channels 20ms
2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 sofia/internal/[hidden email].31 receive message [TRANSCODING_NECESSARY]
2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing file
2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en]
2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message.wav] (en:en)
2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@16000hz 1 channels 20ms
2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 sofia/internal/[hidden email].31 receive message [TRANSCODING_NECESSARY]
2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing file
2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 sofia/internal/[hidden email].31 receive message [TRANSCODING_NECESSARY]
2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate doesn't match
2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec Activated
2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748
2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original codec.
2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less than minimum record length: 3, discarding it.
2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en]
2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-too-small.wav] (en:en)
2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@16000hz 1 channels 20ms
2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 sofia/internal/[hidden email].31 receive message [
the original codec is wideband 16kHz Speex and the wireshark shows that the FS used the same codec. I used FS 1.04 in fedora 8.
I have two questions here,
(1) why does FS report "Sample rate doesn't match"? is it a bug or configuration issue?
(2) Why does FS change the RTP port ? how to fix it?
Thanks,
Regards,
_______________________________________________
FreeSWITCH-users mailing list ============================== you must only have 8k sounds so the resample is when it's playing files
try make hd-sounds-install to install 16k sounds too
============================== Hi, Anthony,
Thanks for your reply.
When I type the command below, I got the error,
Unknown target hd-sound-install
make[1]: *** [hd-sound-install] Error 1
make: *** [hd-sound-install] Error 2
I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should explain why at first FS played in right sample rate. But after playing serveral time, FS complained about sample rate not matching. Any clue? Thanks,
=================================
that was make hd-sounds-install sorrry
you should also update to SVN trunk because based on the line number in your log
its clear you are using a much older version of FS
==============================
==============================
Hi, Anthony and Mike,
Thanks for your reply. The problem still exists even after I ran "make hd-sounds install".
I will try the latest version from the SVN to see if the problem will go away. I will let you know.
Thanks folks,
Regards, =============================== Hi, Anthony and Mike,
With the latest version from SVN, I was able to remove the warning "sample rate not matching". But the remote RTP port was still changed after after playing the vm greeting. See below,
2009-12-03 13:44:46.901216 [INFO] switch_rtp.c:1975 Auto Changing port from XXX.YYY.ZZZ.39:10002 to XXX.YYY.ZZZ.39:3335
Any clue?
I looked at the source code in switch_rtp.c:1975, it shows that if rtp_session->autoadj_tally >= 10, then a rtp port change will happen. Any idea about autoadj_tally and what cause the increase of autoadj_tally ?
Thanks,
=================================
Hi, I solved this issue. the reason is because of the different port number between the the one in SDP and the one in real RTP stream. This is very nice feature.
e
|
