[asterisk-bugs] [JIRA] (ASTERISK-29096) Negotiates g729 but RTP sends g711
David Herselman (JIRA)
noreply at issues.asterisk.org
Fri Sep 25 02:13:36 CDT 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29096?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
David Herselman updated ASTERISK-29096:
---------------------------------------
Attachment: chan_sip_leg.png
Wireshark snippet of the 'A leg' portion of the call. Showing RTP towards calling party being G.711a
> Negotiates g729 but RTP sends g711
> ----------------------------------
>
> Key: ASTERISK-29096
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29096
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/CodecHandling
> Affects Versions: 16.13.0
> Environment: CentOS 7, Asterisk 16.13.0 with upstream (chan_sip) and downstream (chan_iax2)
> Reporter: David Herselman
> Labels: webrtc
> Attachments: chan_iax2_leg.png, chan_sip_leg.png
>
>
> I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it’s from a WebRTC client which doesn’t transmit any audio, could this be influencing Asterisk?
> In the below an incoming call from upstream provider uses chan_sip to receive a call from +27888888888 for 0100000000. Trunk for upstream allows several codecs but first dialplan action looks up destination and forces g729 before sending it on to a downstream chan_iax trunk.
> Both legs of the call negotiate g729 in both directions (all 4 streams), exclusively receives g729 media from downstream calling destination but then unnecessarily transcodes and transmits g711a media to +27888888888, although it wasn't negotiated.
> PS: I was able to confirm with Unsniff that media received from the downstream IAX2 'B' party was encoded with G.729a. Wireshark shows a packet size of 74 bytes, which comprises of 20 bytes of encoded media.
> Wireshark snippet of the SIP portion of the call (upstream -> Asterisk) showing outgoing G711a RTP stream:
> Available here: https://ibb.co/jRGvvVc
> Also uploaded as chan_sip_leg.png
> Wireshark snippet of rhe IAX2 portion of the call (Asterisk -> downstream) showing incoming packets with a size of 74 bytes:
> Available here: https://ibb.co/jRGvvVc
> Also uploaded as chan_iax2_leg.png
> Asterisk debug:
> {code}
> [2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c:
> <--- SIP read from UDP:41.11.11.12:5060 --->
> INVITE sip:0100000000 at 52.22.22.22:5160 SIP/2.0
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>
> Contact: <sip:+27888888888 at 41.11.11.11:5070>
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 102 INVITE
> User-Agent: PortaOne
> Max-Forwards: 69
> Date: Sat, 19 Sep 2020 21:42:19 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> X-LOCATION:
> X-StartID: SDvb9r601-d482cc2b5e0b32417957ea02aaade464-a04aba0
> Content-Type: application/sdp
> Content-Length: 283
> v=0
> o=root 6009 6009 IN IP4 41.11.11.11
> s=session
> c=IN IP4 41.11.11.11
> t=0 0
> m=audio 13918 RTP/AVP 18 8 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> <------------->
> [2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c: --- (18 headers 14 lines) ---
> [2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c: Sending to 41.11.11.12:5060 (NAT)
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Sending to 41.11.11.12:5060 (NAT)
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Using INVITE request as basis request - 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found peer 'Upstream' for '+27888888888' from 41.11.11.12:5060
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] netsock2.c: Using SIP RTP TOS bits 184
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] netsock2.c: Using SIP RTP CoS mark 5
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Got SDP version 6009 and unique parts [root 6009 IN IP4 41.11.11.11]
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found RTP audio format 18
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found RTP audio format 8
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found RTP audio format 101
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found audio description format G729 for ID 18
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found audio description format PCMA for ID 8
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Found audio description format telephone-event for ID 101
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Capabilities: us - (g722|alaw|g729), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] res_rtp_asterisk.c: 0x7f02241bdaf0 -- Strict RTP learning after remote address set to: 41.11.11.11:13918
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Peer audio RTP is at port 41.11.11.11:13918
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Looking for 0100000000 in from-pstn (domain 52.22.22.22)
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] sip/route.c: sip_route_dump: route/path hop: <sip:41.11.11.12;lr;ftag=as40fe2614>
> [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c:
> <--- Transmitting (NAT) to 41.11.11.12:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0;received=41.11.11.12;rport=5060
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 102 INVITE
> Server: Asterisk PBX 16.13.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:0100000000 at 52.22.22.22:5160>
> Content-Length: 0
> <------------>
> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:1] Set("SIP/Upstream-00021a0d", "1?SIP_CODEC=g729") in new stack
> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:2] NoOp("SIP/Upstream-00021a0d", "SIP Call ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11") in new stack
> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000 at incoming:3] Dial("SIP/Upstream-00021a0d", "iax2/Downstream/0100000000") in new stack
> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] app_dial.c: Called iax2/Downstream/0100000000
> [2020-09-19 23:42:20] VERBOSE[2602][C-00021a1f] chan_iax2.c: Call accepted by 196.43.209.105:4569 (format g729)
> [2020-09-19 23:42:20] VERBOSE[2602][C-00021a1f] chan_iax2.c: Format for call is (g729)
> [2020-09-19 23:42:20] VERBOSE[15153][C-00021a1f] app_dial.c: IAX2/Downstream-26055 is ringing
> [2020-09-19 23:42:20] VERBOSE[15153][C-00021a1f] chan_sip.c:
> <--- Transmitting (NAT) to 41.11.11.12:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0;received=41.11.11.12;rport=5060
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 102 INVITE
> Server: Asterisk PBX 16.13.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:0100000000 at 52.22.22.22:5160>
> Content-Length: 0
> <------------>
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] app_dial.c: IAX2/Downstream-26055 answered SIP/Upstream-00021a0d
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] chan_sip.c: Set codec to 'g729' for this call because of ${SIP_CODEC*} variable
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] chan_sip.c: Set codec to 'g729' for this call because of ${SIP_CODEC*} variable
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] chan_sip.c: Audio is at 17678
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] chan_sip.c: Adding codec g729 to SDP
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] chan_sip.c:
> <--- Reliably Transmitting (NAT) to 41.11.11.12:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0;received=41.11.11.12;rport=5060
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 102 INVITE
> Server: Asterisk PBX 16.13.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:0100000000 at 52.22.22.22:5160>
> Content-Type: application/sdp
> Content-Length: 259
> v=0
> o=root 430525994 430525994 IN IP4 52.22.22.22
> s=Asterisk PBX 16.13.0
> c=IN IP4 52.22.22.22
> t=0 0
> m=audio 17678 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=maxptime:230
> a=sendrecv
> <------------>
> [2020-09-19 23:42:22] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel IAX2/Downstream-26055 joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9>
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel SIP/Upstream-00021a0d joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9>
> [2020-09-19 23:42:22] VERBOSE[2637] chan_sip.c:
> <--- SIP read from UDP:41.11.11.12:5060 --->
> ACK sip:0100000000 at 52.22.22.22:5160 SIP/2.0
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.2
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK7caba5c1;rport=5070
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82
> Contact: <sip:+27888888888 at 41.11.11.11:5070>
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 102 ACK
> User-Agent: PortaOne
> Max-Forwards: 69
> Content-Length: 0
> <------------->
> [2020-09-19 23:42:22] VERBOSE[2637] chan_sip.c: --- (12 headers 0 lines) ---
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020640, ts 000160, len 000160)
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020641, ts 000320, len 000160)
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020642, ts 000480, len 000160)
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020643, ts 000640, len 000160)
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020644, ts 000800, len 000160)
> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020645, ts 000960, len 000160)
> <snip>
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020999, ts 250248, len 000160)
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 021000, ts 250408, len 000160)
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 021001, ts 250568, len 000160)
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 021002, ts 250728, len 000160)
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 021003, ts 250888, len 000160)
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 021004, ts 251048, len 000160)
> [2020-09-19 23:42:29] VERBOSE[2637] chan_sip.c:
> <--- SIP read from UDP:41.11.11.12:5060 --->
> BYE sip:0100000000 at 52.22.22.22:5160 SIP/2.0
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK5df7.1435d67.0
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK577fb6bb;rport=5070
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 103 BYE
> User-Agent: PortaOne
> Max-Forwards: 69
> Reason: Q.850 ;cause=16; text="Normal Clearing"
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
> <------------->
> [2020-09-19 23:42:29] VERBOSE[2637] chan_sip.c: --- (13 headers 0 lines) ---
> [2020-09-19 23:42:29] VERBOSE[2637][C-00021a1f] chan_sip.c: Sending to 41.11.11.12:5060 (NAT)
> [2020-09-19 23:42:29] VERBOSE[2637][C-00021a1f] chan_sip.c: Scheduling destruction of SIP dialog '7030be5a09d89a9543234da051897a49 at 41.11.11.11' in 32000 ms (Method: BYE)
> [2020-09-19 23:42:29] VERBOSE[2637][C-00021a1f] chan_sip.c:
> <--- Transmitting (NAT) to 41.11.11.12:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK5df7.1435d67.0;received=41.11.11.12;rport=5060
> Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK577fb6bb;rport=5070
> Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
> From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
> To: <sip:0100000000 at 52.22.22.22:5160>;tag=as11a1cd82
> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.11.11
> CSeq: 103 BYE
> Server: Asterisk PBX 16.13.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
> <------------>
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel SIP/Upstream-00021a0d left 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9>
> [2020-09-19 23:42:29] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel IAX2/Downstream-26055 left 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9>
> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] pbx.c: Spawn extension (incoming, Downstream_0100000000, 1) exited non-zero on 'SIP/Upstream-00021a0d'
> [2020-09-19 23:42:29] VERBOSE[15154][C-00021a1f] chan_iax2.c: Hungup 'IAX2/Downstream-26055'
> {code}
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