Hi, 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?
I couldn't post the asterisk debug in the mailing list, perhaps we could consider increasing maximum messages sizes from 40 KiB in 2020? I subsequently opened a case in JIRA instead (https://issues.asterisk.org/jira/browse/ASTERISK-29096). In the below +27888888888 (chan_sip) calls 0100000000 (chan_iax), negotiates g729 on both legs of the bridged call, exclusively receives g729 media from 0100000000 (chan_iax) but then transmits g711a media to +27888888888 (chan_sip). Herewith the scrubbed and reduced dialogues, full details are available in the JIRA case: [2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c: <--- SIP read from UDP:41.11.11.12:5060 ---> INVITE sip:0100000000@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@41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000@52.22.22.22:5160> Contact: <sip:+27888888888@41.11.11.11:5070> Call-ID: 7030be5a09d89a9543234da051897a49@41.11.11.11<mailto:7030be5a09d89a9543234da051897a49@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 <-------------> <snip> [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) <snip> [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000@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@incoming:2] NoOp("SIP/Upstream-00021a0d", "SIP Call ID: 7030be5a09d89a9543234da051897a49@41.11.11.11<mailto:7030be5a09d89a9543234da051897a49@41.11.11.11>") in new stack [2020-09-19 23:42:19] VERBOSE[15153][C-00021a1f] pbx.c: Executing [0100000000@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) <snip> [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@41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000@52.22.22.22:5160>;tag=as11a1cd82 Call-ID: 7030be5a09d89a9543234da051897a49@41.11.11.11<mailto:7030be5a09d89a9543234da051897a49@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@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: <snip> [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) Regards David Herselman
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