Asterisk Audio quality problem

Ascariota · November 7, 2023, 2:06pm

We have audio issues during calls.

The issue is always the same.
For the first 6-7 seconds of the call, the caller audio is really bad, so that the audio is inaudible.
During the rest of the call, the caller audio is fine but the the recipient audio is not good. It is audible but the quality is low.

Pooh · November 7, 2023, 2:33pm

Basics: which version of Asterisk / which O/S / which version?
Networking: what’s your inbound and outbound Internet connectivity
bandwidth (I don’t mean what does the provider tell you you can “expect”, what
do you actually get)?
Codecs: what are you using, are they same on both sides? Have you tried
changing them?
Detail: You say “caller audio is fine (after a while) but recipient (callee)
is not good”. Does it matter whether the call was “inbound” or “outbound” as
far as your Internet connection is concerned? In other words is it always the
person in your internal network who has good audio, or is it always the person
who places the call?
Have you done a packet capture of SIP + RTP and used any tool such as
wireshark to investigate?

The more information you give us, the more likely it is we can help.

Antony.

david551 · November 7, 2023, 3:15pm

This is very subjective and doesn’t help us understand the likely mechanisms.

Ascariota · November 7, 2023, 6:38pm

1 - Asterisk 20 (we update from 18) on linux physical machine (before virtual).
2 - Network 500 up and down.
3 - G720 (we try other codec but same issues)
4 - The problem occurs even if the load is very low.
5 - Not yet.

We saw that for calls with the problem we have in the logs a lot of Strict RTP learning
Tomorrow we try with Strict RTP disable.

we have the problem in inbound and outbound.
But it is especially on the outbound side that this impacts us.
On inbound it seems that the problem is less present

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– Executing [+3242291649atfrom-sip1trunk:3] Set(“PJSIP/sip_trunk1-00071d2e”, “JITTERBUFFER(adaptive)=default”) in new stack
– Executing [+3242291649atfrom-sip1trunk:4] Set(“PJSIP/sip_trunk1-00071d2e”, “CALLERID(name)=Inbound”) in new stack
– Executing [+3242291649atfrom-sip1trunk:5] Set(“PJSIP/sip_trunk1-00071d2e”, “CDR(DNIS)=+3242291649”) in new stack
– Executing [+3242291649atfrom-sip1trunk:6] Gosub(“PJSIP/sip_trunk1-00071d2e”, “create-ucid,s,1”) in new stack
– Executing [satcreate-ucid:1] Set(“PJSIP/sip_trunk1-00071d2e”, “UCID=”) in new stack
– Executing [satcreate-ucid:2] Set(“PJSIP/sip_trunk1-00071d2e”, “X_PBXID=”) in new stack
– Executing [satcreate-ucid:3] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?4:11”) in new stack
– Goto (create-ucid,s,4)
– Executing [satcreate-ucid:4] Set(“PJSIP/sip_trunk1-00071d2e”, “UID_PART=0000050349”) in new stack
– Executing [satcreate-ucid:5] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?6:8”) in new stack
– Goto (create-ucid,s,6)
– Executing [satcreate-ucid:6] Set(“PJSIP/sip_trunk1-00071d2e”, “__UCID=02000503491699368424”) in new stack
– Executing [satcreate-ucid:7] Goto(“PJSIP/sip_trunk1-00071d2e”, “9”) in new stack
– Goto (create-ucid,s,9)
– Executing [satcreate-ucid:9] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Finish if_if_create-ucid_69_70”) in new stack
– Executing [satcreate-ucid:10] Goto(“PJSIP/sip_trunk1-00071d2e”, “12”) in new stack
– Goto (create-ucid,s,12)
– Executing [satcreate-ucid:12] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Finish if_create-ucid_69”) in new stack
– Executing [satcreate-ucid:13] Set(“PJSIP/sip_trunk1-00071d2e”, “CDR(ucid)=02000503491699368424”) in new stack
– Executing [satcreate-ucid:14] UserEvent(“PJSIP/sip_trunk1-00071d2e”, “SetUCID,UCID: 02000503491699368424”) in new stack
– Executing [satcreate-ucid:15] Return(“PJSIP/sip_trunk1-00071d2e”, “”) in new stack
– Executing [+3242291649atfrom-sip1trunk:7] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?8:13”) in new stack
– Goto (from-sip1trunk,+3242291649,8)
– Executing [+3242291649atfrom-sip1trunk:8] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Incoming call from +3222354000 to +3242291649 matching on complete number”) in new stack
– Executing [+3242291649atfrom-sip1trunk:9] Gosub(“PJSIP/sip_trunk1-00071d2e”, “check-did,s,1(+3242291649,11;100667;pxm_cc)”) in new stack
– Executing [satcheck-did:1] MSet(“PJSIP/sip_trunk1-00071d2e”, “LOCAL(exten)=+3242291649”) in new stack
– Executing [satcheck-did:2] MSet(“PJSIP/sip_trunk1-00071d2e”, “LOCAL(db_result)=11;100667;pxm_cc”) in new stack
– Executing [satcheck-did:3] Set(“PJSIP/sip_trunk1-00071d2e”, “LOCAL(del)=11”) in new stack
– Executing [satcheck-did:4] Set(“PJSIP/sip_trunk1-00071d2e”, “LOCAL(add)=100667”) in new stack
– Executing [satcheck-did:5] Set(“PJSIP/sip_trunk1-00071d2e”, “LOCAL(ctx)=pxm_cc”) in new stack
– Executing [satcheck-did:6] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?7:14”) in new stack
– Goto (check-did,s,7)
– Executing [satcheck-did:7] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?8:9”) in new stack
– Goto (check-did,s,8)
– Executing [satcheck-did:8] Set(“PJSIP/sip_trunk1-00071d2e”, “DEST_EXTEN=”) in new stack
– Executing [satcheck-did:9] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Finish if_if_check-did_71_72”) in new stack
– Executing [satcheck-did:10] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?11:12”) in new stack
– Goto (check-did,s,11)
– Executing [satcheck-did:11] Set(“PJSIP/sip_trunk1-00071d2e”, “DEST_EXTEN=100667”) in new stack
– Executing [satcheck-did:12] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Finish if_if_check-did_71_73”) in new stack
– Executing [satcheck-did:13] Set(“PJSIP/sip_trunk1-00071d2e”, “DEST_CTX=pxm_cc”) in new stack
– Executing [satcheck-did:14] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Finish if_check-did_71”) in new stack
– Executing [satcheck-did:15] Return(“PJSIP/sip_trunk1-00071d2e”, “”) in new stack
– Executing [+3242291649atfrom-sip1trunk:10] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Incoming call from +3222354000 to +3242291649 routing to 100667atpxm_cc”) in new stack
– Executing [+3242291649atfrom-sip1trunk:11] Goto(“PJSIP/sip_trunk1-00071d2e”, “pxm_cc,100667,1”) in new stack
– Goto (pxm_cc,100667,1)
– Executing [100667atpxm_cc:1] NoOp(“PJSIP/sip_trunk1-00071d2e”, “Incoming Proximus CC call from +3222354000 for 100667 1699368424.691969”) in new stack
– Executing [100667atpxm_cc:2] UserEvent(“PJSIP/sip_trunk1-00071d2e”, “SetAccountCode,AccountCode: 9000”) in new stack
– Executing [100667atpxm_cc:3] Set(“PJSIP/sip_trunk1-00071d2e”, “CHANNEL(accountcode)=9000”) in new stack
– Executing [100667atpxm_cc:4] Set(“PJSIP/sip_trunk1-00071d2e”, “STATE=NOT_INUSE”) in new stack
– Executing [100667atpxm_cc:5] Set(“PJSIP/sip_trunk1-00071d2e”, “PRESENCE=unavailable”) in new stack
– Executing [100667atpxm_cc:6] Set(“PJSIP/sip_trunk1-00071d2e”, “CHANNEL(userfield)=100667”) in new stack
– Executing [100667atpxm_cc:7] GotoIf(“PJSIP/sip_trunk1-00071d2e”, “1?8:17”) in new stack
– Goto (pxm_cc,100667,8)
– Executing [100667atpxm_cc:8] Set(“PJSIP/sip_trunk1-00071d2e”, “AUTO_ANSWER=“false””) in new stack
– Executing [100667atpxm_cc:9] Set(“PJSIP/sip_trunk1-00071d2e”, “AUTO_ANSWER_TIME=0”) in new stack
– Executing [100667atpxm_cc:10] Set(“PJSIP/sip_trunk1-00071d2e”, “HOLDABLE_CALL=“false””) in new stack
– Executing [100667atpxm_cc:11] Set(“PJSIP/sip_trunk1-00071d2e”, “TRANSFERABLE_CALL=“false””) in new stack
– Executing [100667atpxm_cc:12] Set(“PJSIP/sip_trunk1-00071d2e”, “DISPLAY_INFO=“Proximus CC””) in new stack
– Executing [100667atpxm_cc:13] Set(“PJSIP/sip_trunk1-00071d2e”, “CHANNEL(musicclass)=silence”) in new stack
– Executing [100667atpxm_cc:14] Gosub(“PJSIP/sip_trunk1-00071d2e”, “record_call,s,1”) in new stack
– Executing [satrecord_call:1] MSet(“PJSIP/sip_trunk1-00071d2e”, “CDR(IsRecorded)=1”) in new stack
– Executing [satrecord_call:2] MixMonitor(“PJSIP/sip_trunk1-00071d2e”, “/records/2023/11/07/02000503491699368424.wav,b”) in new stack
– Executing [satrecord_call:3] Return(“PJSIP/sip_trunk1-00071d2e”, “”) in new stack
== Begin MixMonitor Recording PJSIP/sip_trunk1-00071d2e
– Executing [100667atpxm_cc:15] Dial(“PJSIP/sip_trunk1-00071d2e”, “PJSIP/100667,Irb(set-call-to-agent-headers^s^1(“false”,0,“false”,“false”,“Proximus CC”,02000503491699368424))”) in new stack
– PJSIP/100667-00071d2f Internal Gosub(set-call-to-agent-headers,s,1(false,0,false,false,Proximus CC,02000503491699368424)) start
– Executing [satset-call-to-agent-headers:1] Set(“PJSIP/100667-00071d2f”, “JITTERBUFFER(adaptive)=default”) in new stack
– Executing [satset-call-to-agent-headers:2] GotoIf(“PJSIP/100667-00071d2f”, “1?3:5”) in new stack
– Goto (set-call-to-agent-headers,s,3)
– Executing [satset-call-to-agent-headers:3] NoOp(“PJSIP/100667-00071d2f”, “Setting call to agent Auto-Answer header: false”) in new stack
– Executing [satset-call-to-agent-headers:4] Set(“PJSIP/100667-00071d2f”, “PJSIP_HEADER(add,X-Auto-Answer)=false”) in new stack
– Executing [satset-call-to-agent-headers:5] NoOp(“PJSIP/100667-00071d2f”, “Finish if_set-call-to-agent-headers_63”) in new stack
– Executing [satset-call-to-agent-headers:6] GotoIf(“PJSIP/100667-00071d2f”, “1?7:9”) in new stack
– Goto (set-call-to-agent-headers,s,7)
– Executing [satset-call-to-agent-headers:7] NoOp(“PJSIP/100667-00071d2f”, “Setting call to agent Auto-Answer Time header: 0”) in new stack
– Executing [satset-call-to-agent-headers:8] Set(“PJSIP/100667-00071d2f”, “PJSIP_HEADER(add,X-Auto-Answer-Time)=0”) in new stack
– Executing [satset-call-to-agent-headers:9] NoOp(“PJSIP/100667-00071d2f”, “Finish if_set-call-to-agent-headers_64”) in new stack
– Executing [satset-call-to-agent-headers:10] GotoIf(“PJSIP/100667-00071d2f”, “1?11:13”) in new stack
– Goto (set-call-to-agent-headers,s,11)
– Executing [satset-call-to-agent-headers:11] NoOp(“PJSIP/100667-00071d2f”, “Setting call to agent Holdable Call header: false”) in new stack
– Executing [satset-call-to-agent-headers:12] Set(“PJSIP/100667-00071d2f”, “PJSIP_HEADER(add,X-Holdable-Call)=false”) in new stack
– Executing [satset-call-to-agent-headers:13] NoOp(“PJSIP/100667-00071d2f”, “Finish if_set-call-to-agent-headers_65”) in new stack
– Executing [satset-call-to-agent-headers:14] GotoIf(“PJSIP/100667-00071d2f”, “1?15:17”) in new stack
– Goto (set-call-to-agent-headers,s,15)
– Executing [satset-call-to-agent-headers:15] NoOp(“PJSIP/100667-00071d2f”, “Setting call to agent Transferable Call header: false”) in new stack
– Executing [satset-call-to-agent-headers:16] Set(“PJSIP/100667-00071d2f”, “PJSIP_HEADER(add,X-Transferable-Call)=false”) in new stack
– Executing [satset-call-to-agent-headers:17] NoOp(“PJSIP/100667-00071d2f”, “Finish if_set-call-to-agent-headers_66”) in new stack
– Executing [satset-call-to-agent-headers:18] GotoIf(“PJSIP/100667-00071d2f”, “1?19:21”) in new stack
– Goto (set-call-to-agent-headers,s,19)
– Executing [satset-call-to-agent-headers:19] NoOp(“PJSIP/100667-00071d2f”, “Setting call to agent Display-Info header: Proximus CC”) in new stack
– Executing [satset-call-to-agent-headers:20] Set(“PJSIP/100667-00071d2f”, “PJSIP_HEADER(add,X-Display-Info)=Proximus CC”) in new stack
– Executing [satset-call-to-agent-headers:21] NoOp(“PJSIP/100667-00071d2f”, “Finish if_set-call-to-agent-headers_67”) in new stack
– Executing [satset-call-to-agent-headers:22] GotoIf(“PJSIP/100667-00071d2f”, “1?23:25”) in new stack
– Goto (set-call-to-agent-headers,s,23)
– Executing [satset-call-to-agent-headers:23] NoOp(“PJSIP/100667-00071d2f”, “Setting call to agent UCID header: 02000503491699368424”) in new stack
– Executing [satset-call-to-agent-headers:24] Set(“PJSIP/100667-00071d2f”, “PJSIP_HEADER(add,X-UCID)=02000503491699368424”) in new stack
– Executing [satset-call-to-agent-headers:25] NoOp(“PJSIP/100667-00071d2f”, “Finish if_set-call-to-agent-headers_68”) in new stack
– Executing [satset-call-to-agent-headers:26] Return(“PJSIP/100667-00071d2f”, “0”) in new stack
== Spawn extension (internal, 100667, 1) exited non-zero on ‘PJSIP/100667-00071d2f’
– PJSIP/100667-00071d2f Internal Gosub(set-call-to-agent-headers,s,1(false,0,false,false,Proximus CC,02000503491699368424)) complete GOSUB_RETVAL=0
– Called PJSIP/100667
– Connected line update to PJSIP/sip_trunk1-00071d2e prevented.
– PJSIP/100667-00071d2f is ringing
– Started music on hold, class ‘silence’, on channel ‘PJSIP/635549-00071c96’
– Channel PJSIP/632145-00071d0b left ‘simple_bridge’ basic-bridge
– Channel PJSIP/nomado-trunk-00071d0c left ‘simple_bridge’ basic-bridge
== Spawn extension (to-nomado-trunk, 0032471862125, 1) exited non-zero on ‘PJSIP/632145-00071d0b’
– Executing [hatto-nomado-trunk:1] NoOp(“PJSIP/632145-00071d0b”, “Outbound call from 632145 for accound code 5023 duration 4 seconds completed 1699368404.691925”) in new stack
– Executing [hatto-nomado-trunk:2] GotoIf(“PJSIP/632145-00071d0b”, “0?3:4”) in new stack
– Goto (to-nomado-trunk,h,4)
– Executing [hatto-nomado-trunk:4] NoOp(“PJSIP/632145-00071d0b”, “Finish if_to-nomado-trunk_35”) in new stack
== MixMonitor close filestream (mixed)
== End MixMonitor Recording PJSIP/632145-00071d0b
– Channel PJSIP/sip_trunk1-00071c82 left ‘simple_bridge’ basic-bridge
– Channel PJSIP/637035-00071c83 left ‘simple_bridge’ basic-bridge
== Spawn extension (pxm_cc, 637035, 15) exited non-zero on ‘PJSIP/sip_trunk1-00071c82’
– Executing [hatpxm_cc:1] NoOp(“PJSIP/sip_trunk1-00071c82”, “Incoming Proximus CC call from +3222354000 to 637035 duration 126 seconds completed 1699368294.691730”) in new stack
– Executing [hatpxm_cc:2] GotoIf(“PJSIP/sip_trunk1-00071c82”, “0?3:4”) in new stack
– Goto (pxm_cc,h,4)
– Executing [hatpxm_cc:4] NoOp(“PJSIP/sip_trunk1-00071c82”, “Finish if_pxm_cc_22”) in new stack
== MixMonitor close filestream (mixed)
== End MixMonitor Recording PJSIP/sip_trunk1-00071c82
> 0x7f067c1dfa60 – Strict RTP learning after remote address set to: 172.19.106.71:63806
– PJSIP/100677-00071d28 answered PJSIP/sip_trunk1-00071d27
> 0x7f09ac92a140 – Strict RTP learning after remote address set to: 172.18.33.200:33745
– Channel PJSIP/100677-00071d28 joined ‘simple_bridge’ basic-bridge <9d457b84-b4da-4799-a4ce-30129d9c8e90>
– Channel PJSIP/sip_trunk1-00071d27 joined ‘simple_bridge’ basic-bridge <9d457b84-b4da-4799-a4ce-30129d9c8e90>
> 0x7f067c1dfa60 – Strict RTP learning after ICE completion
> 0x7f067c1dfa60 – Strict RTP learning after remote address set to: 172.19.106.71:63806
> 0x7f067c1dfa60 – Strict RTP switching to RTP target address 172.19.106.71:63806 as source
== WebSocket connection from ‘172.18.21.89:65283’ closed
– Started music on hold, class ‘silence’, on channel ‘PJSIP/100772-00071c75’
> 0x7f09ac92a140 – Strict RTP switching to RTP target address 172.18.33.200:33745 as source
> 0x7f073923dcf0 – Strict RTP learning complete - Locking on source address 172.18.33.200:33725
> 0x7f067c37db30 – Strict RTP learning complete - Locking on source address 10.111.1.94:56953
> 0x7f068074e5b0 – Strict RTP learning complete - Locking on source address 172.18.33.200:33730
> 0x7f07e070aa00 – Strict RTP learning complete - Locking on source address 172.19.106.60:62909
> 0x7f089801e630 – Strict RTP learning after remote address set to: 172.19.106.142:52045
– PJSIP/100675-00071d2b answered PJSIP/sip_trunk1-00071d2a
> 0x7f0a143efef0 – Strict RTP learning after remote address set to: 172.18.33.200:33755
– Channel PJSIP/100675-00071d2b joined ‘simple_bridge’ basic-bridge <5d87dd2e-cee0-4cf8-bd67-87763085f011>
– Channel PJSIP/sip_trunk1-00071d2a joined ‘simple_bridge’ basic-bridge <5d87dd2e-cee0-4cf8-bd67-87763085f011>
> 0x7f089801e630 – Strict RTP learning after ICE completion
> 0x7f089801e630 – Strict RTP learning after remote address set to: 172.19.106.142:52045
> 0x7f07386bf690 – Strict RTP learning complete - Locking on source address 172.18.33.200:33735
> 0x7f089801e630 – Strict RTP switching to RTP target address 172.19.106.142:52045 as source
> 0x7f0aa45e2650 – Strict RTP learning complete - Locking on source address 10.113.1.206:49688
– Stopped music on hold on PJSIP/100241-00071859
> 0x7f0a143efef0 – Strict RTP switching to RTP target address 172.18.33.200:33755 as source
> 0x7f07cfc9e4c0 – Strict RTP learning complete - Locking on source address 10.114.1.50:63965
– PJSIP/nomado-trunk-00071d1d answered PJSIP/633315-00071d1c
– Channel PJSIP/nomado-trunk-00071d1d joined ‘simple_bridge’ basic-bridge <02c7180e-302b-4e2c-8de1-25ea2aefcb4c>
– Channel PJSIP/633315-00071d1c joined ‘simple_bridge’ basic-bridge <02c7180e-302b-4e2c-8de1-25ea2aefcb4c>
> 0x7f09ac55b9b0 – Strict RTP learning complete - Locking on source address 172.18.33.200:33740
> 0x7f0aa8bc98d0 – Strict RTP learning after remote address set to: 10.111.0.210:57475
– PJSIP/635728-00071d2d answered PJSIP/sip_trunk1-00071d2c
> 0x7f07cc4c7580 – Strict RTP learning after remote address set to: 172.18.33.200:33760
– Channel PJSIP/635728-00071d2d joined ‘simple_bridge’ basic-bridge <1d144d19-43e8-4f64-8c79-a1884f753a4c>
– Channel PJSIP/sip_trunk1-00071d2c joined ‘simple_bridge’ basic-bridge <1d144d19-43e8-4f64-8c79-a1884f753a4c>
– Stopped music on hold on PJSIP/635549-00071c96
> 0x7f0aa8bc98d0 – Strict RTP learning after ICE completion
> 0x7f0aa8bc98d0 – Strict RTP learning after remote address set to: 10.111.0.210:57475
> 0x7f0aa8bc98d0 – Strict RTP switching to RTP target address 10.111.0.210:57475 as source
> 0x7f07cc4c7580 – Strict RTP switching to RTP target address 172.18.33.200:33760 as source
> 0x7f0a2c9ef7a0 – Strict RTP learning after remote address set to: 172.19.106.63:54123
– PJSIP/100667-00071d2f answered PJSIP/sip_trunk1-00071d2e
> 0x7f067c61b2f0 – Strict RTP learning after remote address set to: 172.18.33.200:33765
– Channel PJSIP/100667-00071d2f joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/sip_trunk1-00071d2e joined ‘simple_bridge’ basic-bridge
> 0x7f0a2c9ef7a0 – Strict RTP learning after ICE completion
> 0x7f0a2c9ef7a0 – Strict RTP learning after remote address set to: 172.19.106.63:54123
> 0x7f0a2c9ef7a0 – Strict RTP switching to RTP target address 172.19.106.63:54123 as source
– Executing [+3242283774atfrom-sip1trunk:1] NoOp(“PJSIP/sip_trunk1-00071d30”, “Incoming call from +3222354000 to +3242283774 UUI: 1699368430.691973”) in new stack
– Executing [+3242283774atfrom-sip1trunk:2] Log(“PJSIP/sip_trunk1-00071d30”, “NOTICE, Incoming call from +3222354000 to +3242283774 UUI: 1699368430.691973”) in new stack

@david551 During the start of the call, the caller’s voice sounds choppy or has a very low audio level.

We can reproduce the problem by making +/- 7 calls almost simultaneously, and one or two calls have these problems.

We tried to install a gateway (Janus) thinking that it could come from the browser with the webrtct but we still had the problem.

system · December 7, 2023, 6:38pm

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Asterisk Audio quality problem

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