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 From: asterisk-users <asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of David Herselman Sent: Wednesday, 23 September 2020 4:17 PM To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com> Subject: [asterisk-users] Negotiates g729 but RTP contains g711 Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip encapsulation. Most calls surprisingly work, presumably by the caller's system identifying the incoming media as g711, whilst very few callers don't hear the IVR prompt. The downstream is unfortunately not within our control but can't be anything other than Asterisk, considering it's using iax2 in trunk mode. We are running Asterisk 16.13.0, not sure what version the downstream is using. caller -> upstream -> us -> downstream (IVR) Herewith the SIP portion of the call, between upstream and us: Available here: https://ibb.co/jRGvvVc Wireshark unfortunately still cannot dissect iax2 trunk captures though, so I didn't know how to conclusively identify where this problem originates. I do however have a concern that the media we are receiving's packet size (74 bytes) indicates that it is most likely G729. Herewith the IAX2 trunk portion of the call, between us and downstream: Available here: https://ibb.co/r07PkkK ie: We appear to have a reproducible environment where an inbound SIP trunk call sent to a downstream IAX2 trunk negotiates g729 in all 4 streams, receives g729 media from downstream iax2 trunk but then transmits g711a upstream. I'm however struggling with the downstream pcap, to establish what's different about these calls. Trunk config and forwarding structure works the identical way for 50+ other flows on the same host. Regards David Herselman
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