Asterisk
  1. Asterisk
  2. ASTERISK-10762

Distortion in Playback of .gsm files over non-GSM channel

        Details

        • Type: Bug Bug
        • Status: Closed
        • Severity: Minor Minor
        • Resolution: Fixed
        • Affects Version/s: None
        • Target Release Version/s: None
        • Component/s: Codecs/codec_gsm
        • Labels:
          None
        • Source Revision Number:
          89241
        • Mantis ID:
          11243
        • Regression:
          No

          Description

          On one of my Asterisk boxes, I am getting distortion on playback of .gsm sound files. Distortion is heard when transcoding is used to a non-GSM channel. Connecting over a GSM channel is fine.

          The other Asterisk server running the same Asterisk SVN version is OK, so it may be related to dual-processor or OS version.

                    • ADDITIONAL INFORMATION ******

          I have two Asterisk servers, both currently running tip of branches/1.4

          svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4
          svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4
          svn checkout http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4

          One is running Debian "Testing" on a twin-processor Pentium III. The other is running Ubuntu "Feisty Fawn" on a single-processor Pentium III. Both are fully updated to today.

          uname -a:

          Debian (2-processor):
          Linux alambil 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007 i686 GNU/Linux

          Ubuntu (1-processor):
          Linux network 2.6.20-16-server #2 SMP Sun Sep 23 19:57:25 UTC 2007 i686 GNU/Linux

          Playback of .gsm files on the Debian-based server started to be distorted the last time I upgraded Asterisk. There are no errors reported. My SIP phone is a Sipura ATA with alaw preferred.

          Debug is shown below. Should the channel write format change as frequently as it does or is this normal?

          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:4468 sip_alloc: Allocating new SIP dialog for 29a21a37-e0b1061e@158.234.80.17 - INVITE (With RTP)
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:15028 handle_request: **** Received INVITE (5) - Command in SIP INVITE
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:2702 do_setnat: Setting NAT on RTP to Off
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:2707 do_setnat: Setting NAT on VRTP to Off
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:4521 find_call: = Found Their Call ID: 29a21a37-e0b1061e@158.234.80.17 Their Tag 394d336356dee2o0 Our tag: as009b4bc0
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:15028 handle_request: **** Received ACK (6) - Command in SIP ACK
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:2165 __sip_ack: Stopping retransmission on '29a21a37-e0b1061e@158.234.80.17' of Response 101: Match Not Found
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:4521 find_call: = Found Their Call ID: 29a21a37-e0b1061e@158.234.80.17 Their Tag 394d336356dee2o0 Our tag: as009b4bc0
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:15028 handle_request: **** Received INVITE (5) - Command in SIP INVITE
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:2702 do_setnat: Setting NAT on RTP to Off
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:2707 do_setnat: Setting NAT on VRTP to Off
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:5319 process_sdp: T38 state changed to 0 on channel <none>
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:5399 process_sdp: We're settling with these formats: 0x8 (alaw)
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:13708 handle_request_invite: Checking SIP call limits for device 230
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:3140 update_call_counter: Updating call counter for incoming call
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:3957 sip_new: *** Our native formats are 0x8 (alaw)
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:3958 sip_new: *** Joint capabilities are 0x8 (alaw)
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:3959 sip_new: *** Our capabilities are 0x8 (alaw)
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:3960 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:3983 sip_new: This channel will not be able to handle video.
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:8176 build_route: build_route: Contact hop: 230 <sip:230@158.234.80.17:5060>
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:13787 handle_request_invite: SIP/230-081b06b8: New call is still down.... Trying...
          [Nov 14 12:16:39] DEBUG[3865]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/230-081b06b8
          [Nov 14 12:16:39] DEBUG[3996]: pbx.c:1814 pbx_extension_helper: Launching 'Answer'
          – Executing [124@passworded:1] Answer("SIP/230-081b06b8", "") in new stack
          [Nov 14 12:16:39] DEBUG[3863]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 230
          [Nov 14 12:16:39] DEBUG[3996]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/230-081b06b8
          [Nov 14 12:16:39] DEBUG[3863]: chan_sip.c:15655 sip_devicestate: Checking device state for peer 230
          [Nov 14 12:16:39] DEBUG[3996]: chan_sip.c:3611 sip_answer: SIP answering channel: SIP/230-081b06b8
          [Nov 14 12:16:39] DEBUG[3863]: devicestate.c:287 do_state_change: Changing state for SIP/230 - state 1 (Not in use)
          [Nov 14 12:16:39] DEBUG[3996]: chan_sip.c:6592 transmit_response_with_sdp: Setting framing from config on incoming call
          [Nov 14 12:16:39] DEBUG[3881]: app_queue.c:575 handle_statechange: Device 'SIP/230' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
          [Nov 14 12:16:39] DEBUG[3996]: chan_sip.c:6356 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True
          [Nov 14 12:16:39] DEBUG[3996]: chan_sip.c:6357 add_sdp: ** Our prefcodec: 0x0 (nothing)
          [Nov 14 12:16:39] DEBUG[3863]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 230
          [Nov 14 12:16:39] DEBUG[3996]: chan_sip.c:6488 add_sdp: – Done with adding codecs to SDP
          [Nov 14 12:16:39] DEBUG[3863]: chan_sip.c:15655 sip_devicestate: Checking device state for peer 230
          [Nov 14 12:16:39] DEBUG[3996]: channel.c:2553 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=21)
          [Nov 14 12:16:39] DEBUG[3863]: devicestate.c:287 do_state_change: Changing state for SIP/230 - state 1 (Not in use)
          [Nov 14 12:16:39] DEBUG[3996]: chan_sip.c:6533 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw)
          [Nov 14 12:16:39] DEBUG[3881]: app_queue.c:575 handle_statechange: Device 'SIP/230' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
          [Nov 14 12:16:39] DEBUG[3996]: pbx.c:1814 pbx_extension_helper: Launching 'SayUnixTime'
          – Executing [124@passworded:2] SayUnixTime("SIP/230-081b06b8", "|GMT0BST|AdBY 'digits/at' IM") in new stack
          [Nov 14 12:16:39] DEBUG[3996]: say.c:3010 ast_say_date_with_format_en: Parsing A (offset 0) in AdBY 'digits/at' IM
          [Nov 14 12:16:39] DEBUG[3996]: channel.c:3041 set_format: Set channel SIP/230-081b06b8 to write format gsm
          [Nov 14 12:16:39] DEBUG[3996]: rtp.c:2731 ast_rtp_write: Ooh, format changed from unknown to alaw
          [Nov 14 12:16:39] DEBUG[3996]: rtp.c:2748 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160
          [Nov 14 12:16:39] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 160 sample intervals
          – <SIP/230-081b06b8> Playing 'digits/day-3' (language 'en')
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:4521 find_call: = Found Their Call ID: 29a21a37-e0b1061e@158.234.80.17 Their Tag 394d336356dee2o0 Our tag: as56a124e1
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:15028 handle_request: **** Received ACK (6) - Command in SIP ACK
          [Nov 14 12:16:39] DEBUG[3865]: chan_sip.c:2165 __sip_ack: Stopping retransmission on '29a21a37-e0b1061e@158.234.80.17' of Response 102: Match Not Found
          [Nov 14 12:16:40] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 0 sample intervals
          [Nov 14 12:16:40] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 0 sample intervals
          [Nov 14 12:16:40] DEBUG[3996]: channel.c:3041 set_format: Set channel SIP/230-081b06b8 to write format alaw
          [Nov 14 12:16:40] DEBUG[3996]: say.c:3010 ast_say_date_with_format_en: Parsing d (offset 1) in AdBY 'digits/at' IM
          [Nov 14 12:16:40] DEBUG[3996]: channel.c:3041 set_format: Set channel SIP/230-081b06b8 to write format gsm
          [Nov 14 12:16:40] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 160 sample intervals
          – <SIP/230-081b06b8> Playing 'digits/h-14' (language 'en')
          [Nov 14 12:16:41] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 0 sample intervals
          [Nov 14 12:16:41] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 0 sample intervals
          [Nov 14 12:16:41] DEBUG[3996]: channel.c:3041 set_format: Set channel SIP/230-081b06b8 to write format alaw
          [Nov 14 12:16:41] DEBUG[3996]: say.c:3010 ast_say_date_with_format_en: Parsing B (offset 2) in AdBY 'digits/at' IM
          [Nov 14 12:16:41] DEBUG[3996]: channel.c:3041 set_format: Set channel SIP/230-081b06b8 to write format gsm
          [Nov 14 12:16:41] DEBUG[3996]: channel.c:2072 ast_settimeout: Scheduling timer at 160 sample intervals

          1. Text File
            1.2-gsm-gcc4.2.patch
            0.7 kB
            Frank Waller
          2. Text File
            1.4-gsm-gcc4.2.patch
            0.9 kB
            Frank Waller
          3. Text File
            11243-maybe-asm.diff
            0.7 kB
            Jason Parker
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            Activity

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            24 older comments
            Hide
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            Jason Parker added a comment -

            Give this patch a try. There was a somewhat similar issue reported, with a fix provided. This patch does basically the same thing for gsm... ASTERISK-11678

            Disclaimer: I don't know assembly (at all), so I have absolutely no idea whether this works.

            Show
            Jason Parker added a comment - Give this patch a try. There was a somewhat similar issue reported, with a fix provided. This patch does basically the same thing for gsm... ASTERISK-11678 Disclaimer: I don't know assembly (at all), so I have absolutely no idea whether this works.
            Hide
            Permalink
            Jason Parker added a comment -

            Really? Nobody wants to test this patch?

            Show
            Jason Parker added a comment - Really? Nobody wants to test this patch?
            Hide
            Permalink
            Digium Subversion added a comment -

            Repository: asterisk
            Revision: 111856

            U branches/1.4/codecs/gsm/inc/private.h

            ------------------------------------------------------------------------
            r111856 | qwell | 2008-03-28 16:41:11 -0500 (Fri, 28 Mar 2008) | 12 lines

            Allow gsm to compile correctly on x86 with gcc4 optimizations.

            (closes issue ASTERISK-10762)
            Reported by: whiskerp
            Patches:
            11243-maybe-asm.diff uploaded by qwell (license 4)
            Tested by: Seggy (IRC)

            Note: While I did write this patch, I would not have found this if fossil
            had not reported and fixed issue ASTERISK-11678. A huge thanks to him for helping
            to (indirectly) find the problem here.

            ------------------------------------------------------------------------

            http://svn.digium.com/view/asterisk?view=rev&revision=111856

            Show
            Digium Subversion added a comment - Repository: asterisk Revision: 111856 U branches/1.4/codecs/gsm/inc/private.h ------------------------------------------------------------------------ r111856 | qwell | 2008-03-28 16:41:11 -0500 (Fri, 28 Mar 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue ASTERISK-10762 ) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue ASTERISK-11678 . A huge thanks to him for helping to (indirectly) find the problem here. ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=111856
            Hide
            Permalink
            Digium Subversion added a comment -

            Repository: asterisk
            Revision: 111857

            _U trunk/
            U trunk/codecs/gsm/inc/private.h

            ------------------------------------------------------------------------
            r111857 | qwell | 2008-03-28 16:41:34 -0500 (Fri, 28 Mar 2008) | 20 lines

            Merged revisions 111856 via svnmerge from
            https://origsvn.digium.com/svn/asterisk/branches/1.4

            ........
            r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines

            Allow gsm to compile correctly on x86 with gcc4 optimizations.

            (closes issue ASTERISK-10762)
            Reported by: whiskerp
            Patches:
            11243-maybe-asm.diff uploaded by qwell (license 4)
            Tested by: Seggy (IRC)

            Note: While I did write this patch, I would not have found this if fossil
            had not reported and fixed issue ASTERISK-11678. A huge thanks to him for helping
            to (indirectly) find the problem here.

            ........

            ------------------------------------------------------------------------

            http://svn.digium.com/view/asterisk?view=rev&revision=111857

            Show
            Digium Subversion added a comment - Repository: asterisk Revision: 111857 _U trunk/ U trunk/codecs/gsm/inc/private.h ------------------------------------------------------------------------ r111857 | qwell | 2008-03-28 16:41:34 -0500 (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue ASTERISK-10762 ) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue ASTERISK-11678 . A huge thanks to him for helping to (indirectly) find the problem here. ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=111857
            Hide
            Permalink
            Digium Subversion added a comment -

            Repository: asterisk
            Revision: 111858

            _U branches/1.6.0/
            U branches/1.6.0/codecs/gsm/inc/private.h

            ------------------------------------------------------------------------
            r111858 | qwell | 2008-03-28 16:41:55 -0500 (Fri, 28 Mar 2008) | 28 lines

            Merged revisions 111857 via svnmerge from
            https://origsvn.digium.com/svn/asterisk/trunk

            ................
            r111857 | qwell | 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) | 20 lines

            Merged revisions 111856 via svnmerge from
            https://origsvn.digium.com/svn/asterisk/branches/1.4

            ........
            r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines

            Allow gsm to compile correctly on x86 with gcc4 optimizations.

            (closes issue ASTERISK-10762)
            Reported by: whiskerp
            Patches:
            11243-maybe-asm.diff uploaded by qwell (license 4)
            Tested by: Seggy (IRC)

            Note: While I did write this patch, I would not have found this if fossil
            had not reported and fixed issue ASTERISK-11678. A huge thanks to him for helping
            to (indirectly) find the problem here.

            ........

            ................

            ------------------------------------------------------------------------

            http://svn.digium.com/view/asterisk?view=rev&revision=111858

            Show
            Digium Subversion added a comment - Repository: asterisk Revision: 111858 _U branches/1.6.0/ U branches/1.6.0/codecs/gsm/inc/private.h ------------------------------------------------------------------------ r111858 | qwell | 2008-03-28 16:41:55 -0500 (Fri, 28 Mar 2008) | 28 lines Merged revisions 111857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111857 | qwell | 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue ASTERISK-10762 ) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue ASTERISK-11678 . A huge thanks to him for helping to (indirectly) find the problem here. ........ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=111858

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                  Updated:
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