AMR will be worse than PCMU, there are other codecs that are free that would address your concerns. On Mon, May 9, 2016 at 2:45 AM, Pradeepa <pradee@gmail.com> wrote:
Hi Brien,
And also, I wanted to move to AMR, due to bandwidth concern and PCMU codec with wav (ADPCM U-Law, 64.0 Kbps, 8 000 Hz) is not clear on the phone.
Many Thanks.
On Mon, May 9, 2016 at 9:55 AM, Pradeepa <pradee@gmail.com> wrote:
Thanks Brian,
As you suggested, when I removed extension from playback, error disappeared. But still I can't hear any audio.
This is in early media. I tested after answering the call. Both times no sound. Where I have done wrong?
If I remove AMR from Freeswitch prefs, then negotiate with PCMU and play properly wav (u-law) but I want to play AMR.
Now log is as below.
EXECUTE sofia/internal/1000@10.169.112.170 pre_answer() [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 [INFO] mod_dptools.c:1260 Sending early media [DEBUG] sofia_glue.c:5282 Audio Codec Compare [ AMR:120:8000:20:0]/[AMR:96:8000:20:12200] [DEBUG] sofia_glue.c:3190 Set Codec sofia/internal/1000@10.169.112.170 AMR/8000 20 ms 160 samples 12200 bits [DEBUG] switch_core_codec.c:111 sofia/internal/1000@10.169.112.170 Original read codec set to AMR:96 [DEBUG] mod_erlang_event.c:156 Sending event CODEC to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 [DEBUG] sofia_glue.c:5451 Set 2833 dtmf send/recv payload to 101 [DEBUG] mod_erlang_event.c:156 Sending event CODEC to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 [DEBUG] sofia_glue.c:3449 AUDIO RTP [sofia/internal/1000@10.169.112.170] 10.169.112.170 port 22134 -> 192.168.0.174 port 4020 codec: 120 ms: 20 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send payload to 101 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf receive payload to 101 [DEBUG] sofia_glue.c:3749 sofia/internal/1000@10.169.112.170 Set rtp dtmf delay to 40 [NOTICE] sofia_glue.c:4360 Pre-Answer sofia/internal/1000@10.169.112.170! [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_PROGRESS_MEDIA to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 [DEBUG] switch_channel.c:3368 (sofia/internal/1000@10.169.112.170) Callstate Change RINGING -> EARLY [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_CALLSTATE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 [DEBUG] mod_sofia.c:2891 Ring SDP: v=0 o=FreeSWITCH 1462744655 1462744656 IN IP4 10.169.112.170 s=FreeSWITCH c=IN IP4 10.169.112.170 t=0 0 m=audio 22134 RTP/AVP 120 101 a=rtpmap:120 AMR/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
DEBUG] switch_core_session.c:871 Send signal sofia/internal/ 1000@10.169.112.170 [BREAK] DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE_COMPLETE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 DEBUG] switch_core_session.c:1016 Send signal sofia/internal/ 1000@10.169.112.170 [BREAK] DEBUG] sofia.c:5815 Channel sofia/internal/1000@10.169.112.170 entering state [early][183] DEBUG] switch_core_session.c:1151 Send signal sofia/internal/ 1000@10.169.112.170 [BREAK] DEBUG] switch_ivr.c:614 sofia/internal/1000@10.169.112.170 Command Execute sleep(500) EXECUTE sofia/internal/1000@10.169.112.170 sleep(500) DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE_COMPLETE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 DEBUG] mod_erlang_event.c:156 Sending event PRIVATE_COMMAND to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 DEBUG] switch_core_session.c:1151 Send signal sofia/internal/ 1000@10.169.112.170 [BREAK] DEBUG] switch_ivr.c:614 sofia/internal/1000@10.169.112.170 Command Execute playback(abc_test/en/test) EXECUTE sofia/internal/1000@10.169.112.170 playback(abc_test/en/test) DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 INFO] mod_native_file.c:94 Opening File [/usr/local/abc/sounds/en/us/callie/abc_test/en/test.AMR] 8000hz DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms DEBUG] mod_erlang_event.c:156 Sending event PLAYBACK_START to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 DEBUG] switch_ivr_play_say.c:1717 done playing file /usr/local/abc/sounds/en/us/callie/abc_test/en/test.AMR DEBUG] mod_erlang_event.c:156 Sending event PLAYBACK_STOP to attached session 67192af8-159b-11e6-a7fc-eb0baf057465 DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE_COMPLETE to attached session 67192af8-159b-11e6-a7fc-eb0baf057465
Many Thanks.
On Sat, May 7, 2016 at 12:58 AM, Brian West <brian@freeswitch.org> wrote:
For native files, you don't include an extension in your playback, you just say playback /en/test, the file will be test.AMR matching the SDP. And bits != sample rate if thats what you're pointing out.
On Fri, May 6, 2016 at 1:58 PM, Pradeepa <pradee@gmail.com> wrote:
Hi,
I want to play amr file in freeswitch IVR. But I get Invalid file format [amr] error.
Any suggestion to solve this problem?
freeswitch log ------------------- 2016-05-05 22:36:02.151586 [INFO] mod_dptools.c:1260 Sending early media 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [AMR:120:8000:20:0]/[AMR:96:8000:20:12200] 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3190 Set Codec sofia/internal/1000@10.169.112.170 AMR/8000 20 ms 160 samples 12200 bits 2016-05-05 22:36:02.151586 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@10.169.112.170 Original read codec set to AMR:96 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:5451 Set 2833 dtmf send/recv payload to 101 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CODEC to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3449 AUDIO RTP [sofia/internal/1000@10.169.112.170] 10.169.112.170 port 22020 -> 192.168.0.174 port 4006 codec: 120 ms: 20 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CODEC to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send payload to 101 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf receive payload to 101 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3749 sofia/internal/ 1000@10.169.112.170 Set rtp dtmf delay to 40 2016-05-05 22:36:02.151586 [NOTICE] sofia_glue.c:4360 Pre-Answer sofia/internal/1000@10.169.112.170! 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_PROGRESS_MEDIA to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] switch_channel.c:3368 (sofia/internal/1000@10.169.112.170) Callstate Change RINGING -> EARLY 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_CALLSTATE to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] mod_sofia.c:2891 Ring SDP: v=0 o=FreeSWITCH 1462490942 1462490943 IN IP4 10.169.112.170 s=FreeSWITCH c=IN IP4 10.169.112.170 t=0 0 m=audio 22020 RTP/AVP 120 101 a=rtpmap:120 AMR/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
2016-05-05 22:36:02.151586 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/1000@10.169.112.170 [BREAK] 2016-05-05 22:36:02.711592 [DEBUG] switch_core_session.c:1151 Send signal sofia/internal/1000@10.169.112.170 [BREAK] 2016-05-05 22:36:02.731593 [DEBUG] switch_ivr.c:614 sofia/internal/ 1000@10.169.112.170 Command Execute playback(abc_test/en/test.amr) EXECUTE sofia/internal/1000@10.169.112.170 playback(abc_test/en/test.amr) 2016-05-05 22:36:02.731593 [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.731593 [ERR] switch_core_file.c:150 Invalid file format [amr] for [/usr/local/abc/sounds/en/us/callie/abc_test/en/test.amr]!
amr prompt details -------------------------
Audio Format : AMR Format/Info : Adaptive Multi-Rate Format profile : Narrow band Duration : 16s 140ms Bit rate mode : Constant Bit rate : 12.8 Kbps Channel(s) : 1 channel Sampling rate : 8 000 Hz Bit depth : 13 bits Stream size : 25.2 KiB (100%)
Thanks.
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest