change the remote RTP port after sample rate doesnot match
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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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,
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 On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis <davis.erwin@gmail.com> wrote:
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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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,
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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, On 12/2/09, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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
On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis <davis.erwin@gmail.com>wrote:
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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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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 On Wed, Dec 2, 2009 at 2:08 PM, Erwin Davis <davis.erwin@gmail.com> wrote:
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,
On 12/2/09, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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
On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis <davis.erwin@gmail.com>wrote:
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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis <davis.erwin@gmail.com> wrote:
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,
Erwin, As Tony said you've actually got a pretty old installation. If this is in production then I would recommend getting a sandbox machine, install trunk using the quick-and-dirty install, and then update the default config to you specific configuration. Test to make sure it works before you put it into production. :) Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run into any issues that require more real-time conversation. -MC
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, On 12/2/09, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis <davis.erwin@gmail.com>wrote:
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,
Erwin,
As Tony said you've actually got a pretty old installation. If this is in production then I would recommend getting a sandbox machine, install trunk using the quick-and-dirty install, and then update the default config to you specific configuration. Test to make sure it works before you put it into production. :)
Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run into any issues that require more real-time conversation. -MC
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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, On 12/2/09, Erwin Davis <davis.erwin@gmail.com> wrote:
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,
On 12/2/09, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis <davis.erwin@gmail.com>wrote:
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,
Erwin,
As Tony said you've actually got a pretty old installation. If this is in production then I would recommend getting a sandbox machine, install trunk using the quick-and-dirty install, and then update the default config to you specific configuration. Test to make sure it works before you put it into production. :)
Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run into any issues that require more real-time conversation. -MC
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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 On 12/2/09, Erwin Davis <davis.erwin@gmail.com> wrote:
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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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/1003@xxx.yyy.zzz.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,
participants (3)
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Anthony Minessale
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Erwin Davis
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Michael Collins