[asterisk-bugs] [JIRA] Updated: (ASTERISK-18404) out-of-order RTP causes DTMF loss

Matt Jordan (JIRA) noreply at issues.asterisk.org
Thu Jul 19 14:26:21 CDT 2012


     [ https://issues.asterisk.org/jira/browse/ASTERISK-18404?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan updated ASTERISK-18404:
-----------------------------------

    Target Release Version/s: 10.8.0
                              1.8.16.0

> out-of-order RTP causes DTMF loss
> ---------------------------------
>
>                 Key: ASTERISK-18404
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18404
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/RTP
>    Affects Versions: 1.8.4
>         Environment: debian amd64
> {code}
> # dpkg -l | grep asterisk
> ii  asterisk                            1:1.8.4.4~dfsg-2             Open Source Private Branch Exchange (PBX)
> ii  asterisk-config                     1:1.8.4.4~dfsg-2             Configuration files for Asterisk
> ii  asterisk-core-sounds-en-gsm         1.4.19-1                     asterisk PBX sound files - English/gsm
> ii  asterisk-dahdi                      1:1.8.4.4~dfsg-2             DAHDI devices support for the Asterisk PBX
> ii  asterisk-doc                        1:1.6.2.9-2+squeeze2         Source code documentation for Asterisk
> ii  asterisk-modules                    1:1.8.4.4~dfsg-2             loadable modules for the Asterisk PBX
> ii  asterisk-moh-opsound-gsm            2.03-1                       asterisk extra sound files - English/gsm
> ii  asterisk-mysql                      1:1.8.4.4~dfsg-2             MySQL database protocol support for the Asterisk PBX
> ii  asterisk-sounds-extra               1.4.9-1                      Additional sound files for the Asterisk PBX
> ii  asterisk-voicemail                  1:1.8.4.4~dfsg-2             simple voicemail support for the Asterisk PBX
> ii  libasterisk-agi-perl                1.01-2                       Collections of Perl modules to be used with Asterisk PBX AGI
> {code}
>            Reporter: Stephane Chazelas
>            Assignee: Matt Jordan
>            Severity: Critical
>             Fix For: 1.8.16.0, 10.8.0
>
>
> We're in the process of upgrading our phone system from trixbox 1.4 to 1.8 on debian).
> Many DTMF RTP frames from one of our VOIP providers arrive out-of-order. See for instance:
> {code}
>   3.977152 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>   3.977163 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>   3.977171 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF One 1
>   3.977179 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF One 1 (end)
>   4.669046 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Four 4
>   4.669056 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Four 4 (end)
>   4.669065 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Four 4 (end)
>   4.669073 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Four 4 (end)
>   5.162105 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Five 5
>   5.162132 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>   5.162141 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
>   5.162149 195.189.173.27 -> 80.193.213.24 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end)
> {code}
> The 4 packets for "1" above arrived in the 3-4-1-2 sequence, and that digit was ignored by asterisk (only 4 and 5 were transmitted which caused the wrong extension to be called.
> When that happens the DTMF is just discarded by asterisk (and that also happens for 2-1-3-4 sequences which are more common).
> It doesn't occur in trixbox's 1.4.24, so it would be a regression at some point.
> Here is the corresponding rtpdebug output
> {code}
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025103, ts 027992, len 000004)
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025103, ts 027992, len 000004, mark 0, event 00000001, end 1, duration 00800) 
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025104, ts 027992, len 000004)
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025104, ts 027992, len 000004, mark 0, event 00000001, end 1, duration 00800) 
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025101, ts 027992, len 000004)
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025101, ts 027992, len 000004, mark 1, event 00000001, end 0, duration 00000) 
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025102, ts 027992, len 000004)
> [2011-09-01 11:39:41] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025102, ts 027992, len 000004, mark 0, event 00000001, end 1, duration 00800) 
> [...]
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025128, ts 033032, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025128, ts 033032, len 000004, mark 1, event 00000004, end 0, duration 00000) 
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF begin '4' received on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF begin ignored '4' on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025129, ts 033032, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025129, ts 033032, len 000004, mark 0, event 00000004, end 1, duration 00800) 
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF end '4' received on SIP/voipfone-200-00000002, duration 100 ms
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF end passthrough '4' on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025131, ts 033032, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025131, ts 033032, len 000004, mark 0, event 00000004, end 1, duration 00800) 
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025130, ts 033032, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025130, ts 033032, len 000004, mark 0, event 00000004, end 1, duration 00800) 
> [...]
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025143, ts 036952, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025143, ts 036952, len 000004, mark 1, event 00000005, end 0, duration 00000) 
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF begin '5' received on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF begin passthrough '5' on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025144, ts 036952, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025144, ts 036952, len 000004, mark 0, event 00000005, end 1, duration 00800) 
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF end '5' received on SIP/voipfone-200-00000002, duration 100 ms
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF end accepted with begin '5' on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF end '5' detected to have actual duration 0 on the wire, emulation will be triggered on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] DTMF[5944] channel.c: DTMF end '5' has duration 0 but want minimum 80, emulating on SIP/voipfone-200-00000002
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025145, ts 036952, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025145, ts 036952, len 000004, mark 0, event 00000005, end 1, duration 00800) 
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP packet from    195.189.173.27:28704 (type 101, seq 025146, ts 036952, len 000004)
> [2011-09-01 11:39:42] VERBOSE[5944] res_rtp_asterisk.c: Got  RTP RFC2833 from   195.189.173.27:28704 (type 101, seq 025146, ts 036952, len 000004, mark 0, event 00000005, end 1, duration 00800) 
> [2011-09-01 11:39:43] DTMF[5944] channel.c: DTMF end emulation of '5' queued on SIP/voipfone-200-00000002
> {code}
> Sequences were:
> * '1': 3-4-1-2
> * '4': 1-2-4-3
> * '5': 1-2-3-4
> Although the '5' digit is the only one that was received in order, note the note above about duration 0. In that 4 packet sequence for each digit, the "start" packet has duration 0, and the 3 following end ones have duration 800.

--
This message is automatically generated by JIRA.
For more information on JIRA, see: http://www.atlassian.com/software/jira



More information about the asterisk-bugs mailing list