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Summary:ASTERISK-27848: rtp: DTMF Breaks With telephony-event/16000
Reporter:Dominic (ringdom)Labels:patch pjsip
Date Opened:2018-05-10 17:58:50Date Closed:2018-06-12 10:42:12
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Resources/res_rtp_asterisk
Versions:15.4.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:CentOS 7 based FreePBX/SNG 7 Distro running on a KVM based virtual machine Linphone Android and Linphone Windows softphonesAttachments:( 0) ASTERISK-27848.diff
Description:This is my first issue, so apologies if I didn't fill everything out quite right.

I've been running into an issue where RFC4733 DTMF fails (the other side of the call doesn't recognize digits are pressed) when high frequency codecs are offered in the initial SIP INVITE.

If I enable a codec in my Linphone settings that uses a frequency greater than 8000 Hz (e.g. speex, opus, etc.) even if that codec is not negotiated, DTMF fails.

Apparently, Asterisk doesn't support {{telephone-event/16000}}, but simply having that codec present in the INVITE SDP shouldn't prevent {{telephone-event/8000}} from working.

Below are some server logs of both a successful and a failed call. The only change made between the two calls is that in the failed call {{speex/16000}} is offered in the initial INVITE. In both cases, as digits are pressed, they arrive on the agreed channel for {{telephone-event/8000}}, but they are not recognized as RFC2833 on the failed call.

This may have something to do with the channel numbers. In the successful call, {{telephone-event/8000}} is assigned the "standard" 101 channel while in the failed call it is assigned 98. Looking at the Asterisk code base, I saw [this line|https://github.com/asterisk/asterisk/blob/15.4.0/main/rtp_engine.c#L3366] which seems like it could be related, but I don't know the code well enough to know what's going on there.

h1. Successful Call (DTMF worked)

h2. Initial INVITE from Caller

{noformat}
[2018-04-06 14:31:55] VERBOSE[6791] res_pjsip_logger.c: <--- Received SIP request (1322 bytes) from TLS:192.168.1.10:36623 --->
INVITE sip:1101@freepbx.server.com SIP/2.0
Via: SIP/2.0/TLS 192.168.1.10:36623;branch=z9hG4bK.x4Iowo7HJ;rport
From: <sip:1004@freepbx.server.com>;tag=iQuPf4GBP
To: sip:1101@freepbx.server.com
CSeq: 20 INVITE
Call-ID: Al9P0yUtod
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 702
Contact: <sip:1004@192.168.1.10:36623;transport=tls>;+sip.instance="<urn:uuid:5a3150e3-5e90-4984-b12b-2cfc5ad4a15e>"
User-Agent: LinphoneAndroid/3.3.2.2-TG (belle-sip/1.6.3)

v=0
o=1004 3003 1159 IN IP4 192.168.1.10
s=Talk
c=IN IP4 192.168.1.10
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/SAVP 96 0 8 18 101
a=rtpmap:96 speex/8000
a=fmtp:96 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Yl6b4HKe7GanfgObzlZmCS3TX1iZLmOMW57JJjJ2
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:gkfj8LHaB1PZyV8LLdGVDnMu88YT1FxZPwcgcwL8
a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:kE1BFsEgifzQw9LAyyD80aaIMeFrOlwf1vA4T44PsKS7tlEAHLF552MJ5uGPKg==
a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:rXGPaoCCvjC0KuwA0aG3HgmHFCGMd5qGeTfGZKYf2WPxhHWKbRYc7MVlrYOsug==
a=rtcp-fb:* ccm tmmbr
{noformat}

h2. Notify Caller of Answer
{noformat}
[2018-04-06 14:31:56] VERBOSE[32423] res_pjsip_logger.c: <--- Transmitting SIP response (1040 bytes) to TLS:192.168.1.10:36623 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.10:36623;rport=36623;received=192.168.1.10;branch=z9hG4bK.g2CzEVYhu
Call-ID: Al9P0yUtod
From: <sip:1004@freepbx.server.com>;tag=iQuPf4GBP
To: <sip:1101@freepbx.server.com>;tag=30828a9d-cdbf-479a-9016-b48215b31e8c
CSeq: 21 INVITE
Server: FPBX-14.0.2.10(15.2.2)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:192.168.1.5:5061;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "User 1" <sip:1101@freepbx.server.com>
Content-Type: application/sdp
Content-Length:   387

v=0
o=- 3003 1161 IN IP4 192.168.1.5
s=Asterisk
c=IN IP4 192.168.1.5
t=0 0
m=audio 12880 RTP/SAVP 0 8 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:k7d54Fzm1HMFS2WU3xr0zyepyrShTeTIqa03aofs
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
{noformat}

h2. RTP Stream

{noformat}
[2018-04-06 14:31:57] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004392, ts 565097400, len 000170)
[2018-04-06 14:31:57] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000005, ts 565097565, len 000160)
[2018-04-06 14:31:57] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004393, ts 565097560, len 000170)
[2018-04-06 14:31:57] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000006, ts 565097725, len 000160)
[2018-04-06 14:31:57] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004394, ts 565097720, len 000170)
[2018-04-06 14:31:57] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000007, ts 565097885, len 000160)
.....
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 101, seq 000092, ts 602617351, len 000004)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP RFC2833 from   192.168.1.10:7076 (type 101, seq 000092, ts 602617351, len 000004, mark 1, event 00000008, end 0, duration 00160)
[2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4)
[2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000110, ts 565114365, len 000160)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004498, ts 565114360, len 000170)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 101, seq 000093, ts 602617351, len 000004)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP RFC2833 from   192.168.1.10:7076 (type 101, seq 000093, ts 602617351, len 000004, mark 0, event 00000008, end 0, duration 00320)
[2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4)
[2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000111, ts 565114525, len 000160)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004499, ts 565114520, len 000170)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 101, seq 000094, ts 602617351, len 000004)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP RFC2833 from   192.168.1.10:7076 (type 101, seq 000094, ts 602617351, len 000004, mark 0, event 00000008, end 0, duration 00480)
[2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4)
[2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000112, ts 565114685, len 000160)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004500, ts 565114680, len 000170)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 101, seq 000095, ts 602617351, len 000004)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got  RTP RFC2833 from   192.168.1.10:7076 (type 101, seq 000095, ts 602617351, len 000004, mark 0, event 00000008, end 0, duration 00640)
[2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4)
[2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000113, ts 565114845, len 000160)
[2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 004501, ts 565114840, len 000170)
{noformat}

h1. Failed Call (DTMF didn't work)

h2. Initial INVITE from Caller

{noformat}
[2018-04-06 14:32:19] VERBOSE[6791] res_pjsip_logger.c: <--- Received SIP request (1406 bytes) from TLS:192.168.1.10:36623 --->
INVITE sip:1101@freepbx.server.com SIP/2.0
Via: SIP/2.0/TLS 192.168.1.10:36623;branch=z9hG4bK.09oNseORt;rport
From: <sip:1004@freepbx.server.com>;tag=0bVmil7L7
To: sip:1101@freepbx.server.com
CSeq: 20 INVITE
Call-ID: giaAPEO5oz
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 786
Contact: <sip:1004@192.168.1.10:36623;transport=tls>;+sip.instance="<urn:uuid:5a3150e3-5e90-4984-b12b-2cfc5ad4a15e>"
User-Agent: LinphoneAndroid/3.3.2.2-TG (belle-sip/1.6.3)

v=0
o=1004 1441 2486 IN IP4 192.168.1.10
s=Talk
c=IN IP4 192.168.1.10
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/SAVP 96 97 0 8 18 101 98
a=rtpmap:96 speex/16000
a=fmtp:96 vbr=on
a=rtpmap:97 speex/8000
a=fmtp:97 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/16000
a=rtpmap:98 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SlMxT019CunqOGrAXwXreRhm6kJ9dw8X3Gh6/LhV
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:pWWTCwxlIGi20WcvZCAJC0Jo29dIrJ/9VmDuk0Ig
a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:PkYK/oG2F8U53pE6Y/1gu05TqqxgqsfFmEegunUOyMsTpNPA8gAEE7QXULB16Q==
a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:US4TIkQ3sl1XK28C7LZPBMGzJBaCGJqxMVt3sg2PBZFIcwv3KomV8ZWZ3DyvRw==
a=rtcp-fb:* ccm tmmbr
{noformat}

h2. Notify Caller of Answer

{noformat}
[2018-04-06 14:32:21] VERBOSE[30966] res_pjsip_logger.c: <--- Transmitting SIP response (1037 bytes) to TLS:192.168.1.10:36623 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.10:36623;rport=36623;received=192.168.1.10;branch=z9hG4bK.MANUueCbB
Call-ID: giaAPEO5oz
From: <sip:1004@freepbx.server.com>;tag=0bVmil7L7
To: <sip:1101@freepbx.server.com>;tag=1add4390-017a-4856-9973-9946756bba26
CSeq: 21 INVITE
Server: FPBX-14.0.2.10(15.2.2)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:192.168.1.5:5061;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "User 1" <sip:1101@freepbx.server.com>
Content-Type: application/sdp
Content-Length:   384

v=0
o=- 1441 2488 IN IP4 192.168.1.5
s=Asterisk
c=IN IP4 192.168.1.5
t=0 0
m=audio 16570 RTP/SAVP 0 8 18 98
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:oVH9HZ2j56kvQFRcwOdlEi550fdfu7fBr9RhTHRN
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
{noformat}

h2. RTP Stream

{noformat}
[2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000004, ts 3208992109, len 000160)
[2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019276, ts 3208992104, len 000170)
[2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000005, ts 3208992269, len 000160)
[2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019277, ts 3208992264, len 000170)
[2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000006, ts 3208992429, len 000160)
[2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019278, ts 3208992424, len 000170)
[2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000007, ts 3208992589, len 000160)
[2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019279, ts 3208992584, len 000170)
[2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000008, ts 3208992749, len 000160)
[2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019280, ts 3208992744, len 000170)
.......
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000078, ts 4113917409, len 000004)
[2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000096, ts 3209006829, len 000160)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019368, ts 3209006824, len 000170)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000079, ts 4113917409, len 000004)
[2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000097, ts 3209006989, len 000160)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019369, ts 3209006984, len 000170)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000080, ts 4113917409, len 000004)
[2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000098, ts 3209007149, len 000160)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019370, ts 3209007144, len 000170)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000081, ts 4113917409, len 000004)
[2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000099, ts 3209007309, len 000160)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019371, ts 3209007304, len 000170)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000082, ts 4113917409, len 000004)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000082, ts 4113917409, len 000014)
[2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000082, ts 4113917409, len 000014)
........
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000124, ts 4113924369, len 000004)
[2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000139, ts 3209013709, len 000160)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019411, ts 3209013704, len 000170)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000125, ts 4113924369, len 000004)
[2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000140, ts 3209013869, len 000160)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019412, ts 3209013864, len 000170)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000126, ts 4113924369, len 000004)
[2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000141, ts 3209014029, len 000160)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019413, ts 3209014024, len 000170)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000127, ts 4113924369, len 000004)
[2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.20:48038 (type 00, seq 000142, ts 3209014189, len 000160)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to      192.168.1.10:7076 (type 00, seq 019414, ts 3209014184, len 000170)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000128, ts 4113924369, len 000004)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000128, ts 4113924369, len 000014)
[2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got  RTP packet from    192.168.1.10:7076 (type 98, seq 000128, ts 4113924369, len 000014)
{noformat}
Comments:By: Asterisk Team (asteriskteam) 2018-05-10 17:58:51.774-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Joshua C. Colp (jcolp) 2018-05-15 02:39:51.307-0500

Can you please attach all the information here instead of linking to the forum? For those who work on issues it makes it easier to keep everything together, and also if people in the future want to reference back.
By: Dominic (ringdom) 2018-05-15 12:23:45.798-0500

Good idea. I added the server logs and other relevant information to the bug description. Let me know if there's anything else I can do to help.
By: Joshua C. Colp (jcolp) 2018-05-31 04:55:48.401-0500

I adapted an existing test to test this scenario and under PJSIP it was still able to receive DTMF as expected even with the presence of telephone-event/16000.

Can you please also provide a packet capture and a full debug level (debug set to go to a file in logger.conf and core set debug 9 done on CLI) showing this?
By: Joshua C. Colp (jcolp) 2018-05-31 11:11:10.877-0500

Aha, figured out the difference and have reproduced it.
By: Dominic (ringdom) 2018-05-31 13:23:33.360-0500

Thanks for looking into this Joshua, I appreciate it. Let me know if you need anything else from my end.
By: Joshua C. Colp (jcolp) 2018-06-05 04:39:41.925-0500

I've attached a patch which I believe should resolve this problem. Can you please apply and try it?
By: Dominic (ringdom) 2018-06-05 14:17:20.848-0500

Thanks Joshua! Give me a day or two to setup a build environment, and I'll let you know how the test goes.
By: Dominic (ringdom) 2018-06-11 23:54:44.431-0500

I was finally able to get a non-production environment setup to test this, and it seems to be working perfectly in my ad hoc testing. I was able to reproduce the issue in the default Asterisk 15 build installed with FreePBX, and then when I built from source with your patch, I was no longer able to reproduce it.

For good measure, I even went back and rebuilt the source without your patch and the problem came right back. I think you nailed it.

As an interesting side note, I was not able to reproduce the issue on Asterisk 13.
By: Friendly Automation (friendly-automation) 2018-06-12 10:42:13.102-0500

Change 9176 merged by Jenkins2:
rtp: Don't negotiate dynamic codecs using payload.

[https://gerrit.asterisk.org/9176|https://gerrit.asterisk.org/9176]
By: Friendly Automation (friendly-automation) 2018-06-12 10:42:52.863-0500

Change 9177 merged by Jenkins2:
rtp: Don't negotiate dynamic codecs using payload.

[https://gerrit.asterisk.org/9177|https://gerrit.asterisk.org/9177]