Infinite codec switching

Asterisk Asterisk Support
1

Following the severe audio quality issues I described at Choppy audio with DSP warnings, we have finally been able to run RTP debug logging during an affected call, and identified the issue.

The issue is that both Asterisk and our provider’s media server appear to switch codecs when they receive a packet using a different codec from the other side. Because this is happening on both ends, some calls get into an infinite loop of switching codecs.

Example call - INVITE SDP (from us):

v=0
o=- 1014299248 1014299248 IN IP4 [our IP]
s=Asterisk
c=IN IP4 [our IP]
t=0 0
m=audio 15968 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

Note that G722 is the first-listed codec.

Example call - 200 SDP (from our provider):

v=0
o=- 1014299248 1014299250 IN IP4 [our provider's IP]
s=session
c=IN IP4 [our provider's IP]
t=0 0
m=audio 16476 RTP/AVP 8 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Note that PCMA is the first-listed codec.

Example call - RTP debug logs:

Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005417, ts 000160, len 000160)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011309, ts 3885110459, len 000170)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006061, ts 062560, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005418, ts 000320, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006062, ts 062720, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011310, ts 3885110619, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005419, ts 000480, len 000160)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011311, ts 3885110779, len 000170)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006063, ts 062880, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005420, ts 000640, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006064, ts 063040, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005421, ts 000800, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006065, ts 063200, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011312, ts 3885110939, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011313, ts 3885111099, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 08, seq 004688, ts 000056, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006066, ts 063360, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 08, seq 005422, ts 000960, len 000160)
Got  RTP packet from    [our provider's IP]:16476 (type 09, seq 004689, ts 000216, len 000160)
[2024-02-19 06:26:35] WARNING[15003]: dsp.c:1469 ast_dsp_silence_noise_with_energy: Can only calculate silence on signed-linear, alaw or ulaw frames :(
Got  RTP packet from    [our IP]:39125 (type 00, seq 011314, ts 3885111259, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005423, ts 001120, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006067, ts 063520, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011315, ts 3885111419, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 09, seq 004690, ts 000376, len 000160)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005424, ts 001280, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006068, ts 063680, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 09, seq 004691, ts 000536, len 000160)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005425, ts 001440, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006069, ts 063840, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 09, seq 004692, ts 000696, len 000160)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011316, ts 3885111579, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011317, ts 3885111739, len 000170)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005426, ts 001600, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006070, ts 064000, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 09, seq 004693, ts 000856, len 000160)
Sent RTP packet to      [our provider's IP]:16476 (type 09, seq 005427, ts 001760, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006071, ts 064160, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011318, ts 3885111899, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011319, ts 3885112059, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 08, seq 004694, ts 001016, len 000160)
Sent RTP packet to      [our provider's IP]:16476 (type 08, seq 005428, ts 001920, len 000160)
Sent RTP packet to      [our IP]:39125 (via ICE) (type 00, seq 006072, ts 064320, len 000170)
Got  RTP packet from    [our IP]:39125 (type 00, seq 011320, ts 3885112219, len 000170)
Got  RTP packet from    [our provider's IP]:16476 (type 09, seq 004695, ts 001176, len 000160)
[2024-02-19 06:26:35] WARNING[15003]: dsp.c:1469 ast_dsp_silence_noise_with_energy: Can only calculate silence on signed-linear, alaw or ulaw frames :(
...

Note that the first packet we send to our provider is G722 (type 09), while the first packet we receive from our provider is PCMA (type 08). Due to receiving a PCMA packet, we switch to PCMA, but at the same time, due to receiving a G722 packet, our provider switches to G722. This switching back and forth goes on forever because both ends are “polite” and willing to switch to the codec used by the other side.

As a temporary/interim measure, we have changed our allowed codec list to only PCMA, so that there is no room for confusion. However, going forward, we would want to use the preferred codec indicated by our provider for a given call where possible, especially to use G722 (wideband) audio when it makes sense (our provider lists codecs in a different order for different calls, presumably indicating order of preference for that call).

What are the best ways to handle this situation? Given that our provider switches codecs on the fly, should we be configuring Asterisk to not do the same? In particular, would it be possible to configure Asterisk to just use the codec preferred by our provider (PCMA in this example) from the start, rather than initially trying to use our first choice codec (G722) and then switching on the fly if we receive something different?

2

This option[1] would disable the automatic switching.

[1] asterisk/configs/samples/pjsip.conf.sample at master · asterisk/asterisk · GitHub

3

Thanks jcolp. I did see that option, but also another forum post where it was suggested this isn’t usually a great idea: Asymmetric_rtp_codec=yes usage

It still could be a good option here, but my only concern is that in practice it might basically be the same as only supporting one codec in the first place, given that Asterisk starts out sending RTP using its own first choice of codec, and then this causes the remote end to switch to that codec anyway.

Do you know if it would be possible to configure Asterisk to start out using the first codec in the 200 response, rather than starting out using its own first choice? Or is it too early to know what’s in the 200 response by the time we start sending RTP?

4

What you see in the documentation is what there is for configurability. I’m not aware of anything specifically for that.

5

Makes sense, thank you. Just so my understanding is clear, is the codec list from the 200 response used in any way by Asterisk?

Presumably it would at least be used to narrow down which codecs can be used (in case any supported by us are not supported by the other side), but just can’t be used to prioritise one codec (that is supported by both sides) over another?

6

It scopes the codecs used, and the first one in the list is used for preference.

7

The first one in which list sorry, the list in our INVITE or in the other side’s 200?

In the example in my original post Asterisk appears to start out sending G722 (the first codec listed in our INVITE) even though the first codec listed in the other side’s 200 is PCMA. That’s why I assumed Asterisk wasn’t using the order in the other side’s 200 for priority.

8

The other side’s 200. Enabling core debug will show information on codec negotiation, choices, etc. That would be needed for any further clarification alongside the complete SIP trace, not just specific messages. For example if the provider sent a 183, that alters things too.

9

Thanks jcolp, complete SIP trace of an affected call below (note that each message appears multiple times due to how our tracing is set up).

I don’t see any 183 response in the SIP trace but perhaps something else will be of note.

Asterisk debug logs will be in the next post.

proto:TCP 2024-02-22T22:20:56.755861Z  10.150.0.2:34717 ---> 10.150.0.32:5165

INVITE sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Contact: <sip:+[from number]@10.150.0.2:6001;transport=TCP>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 589335772 589335772 IN IP4 [provider IP]
s=Asterisk
c=IN IP4 [provider IP]
t=0 0
m=audio 15054 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:TCP 2024-02-22T22:20:56.756568Z  10.150.0.2:34717 ---> 0.0.0.0:5165

INVITE sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Contact: <sip:+[from number]@10.150.0.2:6001;transport=TCP>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 589335772 589335772 IN IP4 [provider IP]
s=Asterisk
c=IN IP4 [provider IP]
t=0 0
m=audio 15054 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:TCP 2024-02-22T22:20:56.757Z  10.150.0.32:5165 ---> 10.150.0.2:34717

SIP/2.0 100 Giving it a try
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
To: <sip:+[dialled number]@10.150.0.32>
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.757209Z  0.0.0.0:5165 ---> 10.150.0.2:34717

SIP/2.0 100 Giving it a try
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
To: <sip:+[dialled number]@10.150.0.32>
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.76601Z  0.0.0.0:42707 ---> 81.201.89.110:5060

INVITE sip:+[dialled number]@[provider outbound domain] SIP/2.0
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK9e73.2f1e0907.0;i=5ca13bc2
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Contact: <sip:+[from number]@[our domain]:5671;transport=tcp;thinfo=VG99bzAdIFt4ZWBcYld7RWlTYhNlX35edkR+UX1hbllgX3JPJBMyPScfPx03SQQiA0NUGzMfeUR+UX1jel9qWnJCZQ-->
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 69
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 589335772 589335772 IN IP4 [provider IP]
s=Asterisk
c=IN IP4 [provider IP]
t=0 0
m=audio 15054 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:TCP 2024-02-22T22:20:56.771Z  10.150.0.32:5165 ---> 10.150.0.2:34717

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Proxy-Authenticate: [redacted]
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.771137Z  0.0.0.0:42707 ---> 81.201.89.110:5060

ACK sip:+[dialled number]@[provider outbound domain] SIP/2.0
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK9e73.2f1e0907.0
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
CSeq: 9331 ACK
Max-Forwards: 70
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.771285Z  81.201.89.110:5060 ---> 0.0.0.0:42707

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK9e73.2f1e0907.0;i=5ca13bc2;received=34.151.108.130
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Proxy-Authenticate: [redacted]
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.771671Z  0.0.0.0:5165 ---> 10.150.0.2:34717

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 INVITE
Proxy-Authenticate: [redacted]
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.772328Z  10.150.0.2:34717 ---> 10.150.0.32:5165

ACK sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 ACK
Max-Forwards: 70
Content-Length:  0


proto:TCP 2024-02-22T22:20:56.772458Z  10.150.0.2:34717 ---> 0.0.0.0:5165

ACK sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 ACK
Max-Forwards: 70
Content-Length:  0


proto:TCP 2024-02-22T22:20:56.77252Z  10.150.0.2:34717 ---> 10.150.0.32:5165

INVITE sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Contact: <sip:+[from number]@10.150.0.2:6001;transport=TCP>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
Proxy-Authorization: [redacted]
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 589335772 589335772 IN IP4 [provider IP]
s=Asterisk
c=IN IP4 [provider IP]
t=0 0
m=audio 15054 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:TCP 2024-02-22T22:20:56.772584Z  10.150.0.2:34717 ---> 0.0.0.0:5165

ACK sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPjdef8d951-4799-4cdc-afe3-599548d2f300;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=68203133343f8300c61e54b3203afddc-645182b5
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9331 ACK
Max-Forwards: 69
Content-Length:  0


proto:TCP 2024-02-22T22:20:56.77285Z  10.150.0.2:34717 ---> 0.0.0.0:5165

INVITE sip:+[dialled number]@10.150.0.32:5165 SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Contact: <sip:+[from number]@10.150.0.2:6001;transport=TCP>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
Proxy-Authorization: [redacted]
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 589335772 589335772 IN IP4 [provider IP]
s=Asterisk
c=IN IP4 [provider IP]
t=0 0
m=audio 15054 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:TCP 2024-02-22T22:20:56.773Z  10.150.0.32:5165 ---> 10.150.0.2:34717

SIP/2.0 100 Giving it a try
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
To: <sip:+[dialled number]@10.150.0.32>
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 INVITE
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.773263Z  0.0.0.0:5165 ---> 10.150.0.2:34717

SIP/2.0 100 Giving it a try
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
To: <sip:+[dialled number]@10.150.0.32>
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 INVITE
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.782576Z  0.0.0.0:42707 ---> 81.201.89.110:5060

INVITE sip:+[dialled number]@[provider outbound domain] SIP/2.0
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK6e73.f4da6b55.0;i=5ca13bc2
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Contact: <sip:+[from number]@[our domain]:5671;transport=tcp;thinfo=VG99bzAdIFt4ZWBcYld7RWlTYhNlX35edkR+UX1hbllgX3JPJBMyPScfPx03SQQiA0NUGzMfeUR+UX1jel9qWnJCZQ-->
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 69
Proxy-Authorization: [redacted]
Content-Type: application/sdp
Content-Length:   283

v=0
o=- 589335772 589335772 IN IP4 [provider IP]
s=Asterisk
c=IN IP4 [provider IP]
t=0 0
m=audio 15054 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:TCP 2024-02-22T22:20:56.798736Z  81.201.89.110:5060 ---> 0.0.0.0:42707

SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK6e73.f4da6b55.0;i=5ca13bc2;received=34.151.108.130
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 INVITE
Content-Length: 0


proto:TCP 2024-02-22T22:20:56.937Z  10.150.0.32:5165 ---> 10.150.0.2:34717

SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9332 INVITE
Contact: <sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU->
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Content-Length:  0


proto:TCP 2024-02-22T22:20:56.937288Z  81.201.89.110:5060 ---> 0.0.0.0:42707

SIP/2.0 180 Ringing
Via: SIP/2.0/TCP [our domain]:5671;received=34.151.108.130;branch=z9hG4bK6e73.f4da6b55.0;i=5ca13bc2
Record-Route: <sip:185.47.148.110:5060;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Record-Route: <sip:185.47.148.110;transport=tcp;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9332 INVITE
Contact: <sip:185.47.148.25:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Content-Length:  0


proto:TCP 2024-02-22T22:20:56.937608Z  0.0.0.0:5165 ---> 10.150.0.2:34717

SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9332 INVITE
Contact: <sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU->
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Content-Length:  0


proto:TCP 2024-02-22T22:21:03.068922Z  81.201.89.110:5060 ---> 0.0.0.0:42707

SIP/2.0 200 OK
Via: SIP/2.0/TCP [our domain]:5671;received=34.151.108.130;branch=z9hG4bK6e73.f4da6b55.0;i=5ca13bc2
Record-Route: <sip:185.47.148.110:5060;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Record-Route: <sip:185.47.148.110;transport=tcp;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9332 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Contact: <sip:185.47.148.25:5060>
Supported: timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   268

v=0
o=- 589335772 589335774 IN IP4 185.47.148.25
s=session
c=IN IP4 185.47.148.25
t=0 0
m=audio 16638 RTP/AVP 8 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

proto:TCP 2024-02-22T22:21:03.069Z  10.150.0.32:5165 ---> 10.150.0.2:34717

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9332 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Contact: <sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU->
Supported: timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   268

v=0
o=- 589335772 589335774 IN IP4 185.47.148.25
s=session
c=IN IP4 185.47.148.25
t=0 0
m=audio 16638 RTP/AVP 8 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

proto:TCP 2024-02-22T22:21:03.069201Z  0.0.0.0:5165 ---> 10.150.0.2:34717

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj154a5c20-3068-4cbc-82c6-a886ac9fd5ac;alias
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9332 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Contact: <sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU->
Supported: timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   268

v=0
o=- 589335772 589335774 IN IP4 185.47.148.25
s=session
c=IN IP4 185.47.148.25
t=0 0
m=audio 16638 RTP/AVP 8 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

proto:TCP 2024-02-22T22:21:03.070474Z  10.150.0.2:34717 ---> 10.150.0.32:5165

ACK sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU- SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPj8e610699-000d-4ea0-8792-0b25f1885df1;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 ACK
Max-Forwards: 70
Content-Length:  0


proto:TCP 2024-02-22T22:21:03.070621Z  10.150.0.2:34717 ---> 0.0.0.0:5165

ACK sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU- SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPj8e610699-000d-4ea0-8792-0b25f1885df1;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 ACK
Max-Forwards: 70
Content-Length:  0


proto:TCP 2024-02-22T22:21:03.071102Z  0.0.0.0:45695 ---> 185.47.148.110:5060

ACK sip:185.47.148.25:5060 SIP/2.0
Route: <sip:185.47.148.110;transport=tcp;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>,<sip:185.47.148.110:5060;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK6e73.f4da6b55.2;i=5ca13bc2
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9332 ACK
Max-Forwards: 69
Content-Length:  0


proto:TCP 2024-02-22T22:21:12.683574Z  10.150.0.2:34717 ---> 10.150.0.32:5165

BYE sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU- SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPj3dcddde1-9306-489f-8a54-55dd74ebf888;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9333 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
Content-Length:  0


proto:TCP 2024-02-22T22:21:12.684191Z  10.150.0.2:34717 ---> 0.0.0.0:5165

BYE sip:10.150.0.32:5165;transport=tcp;thinfo=9m9sHCoEalBrZnpbZ0FyQGhPYmJkVCQdIhojETwhIFIkDDNPPBNoIWZSPwF4EiQANG43CTJbJUdoUH42N1xjQndGNgB+MTBXYEJwQmZXZmA2DDMOIkBuTW8gPR9qXntBflVkfWVbaEFyRWBbZmNiX2sDMU8iU248OlQ2GyITbQI1MWAJY1dyWTUCYGB5W2IJIlkyBWtjeVxmWXVBYwMwMDUOZFFVdCMII2llV2VBd0N+UGdrel1lVXZEZlFDUyAMIFVzWmBPY31kVWVedEU- SIP/2.0
Via: SIP/2.0/TCP 10.150.0.2:6001;rport;branch=z9hG4bKPj3dcddde1-9306-489f-8a54-55dd74ebf888;alias
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9333 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
Content-Length:  0


proto:TCP 2024-02-22T22:21:12.684553Z  0.0.0.0:45695 ---> 185.47.148.110:5060

BYE sip:185.47.148.25:5060 SIP/2.0
Route: <sip:185.47.148.110;transport=tcp;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>,<sip:185.47.148.110:5060;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Via: SIP/2.0/TCP [our domain]:5671;branch=z9hG4bK7e73.42822642.0;i=5ca13bc2
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
CSeq: 9333 BYE
Reason: Q.850;cause=16
Max-Forwards: 69
Content-Length:  0


proto:TCP 2024-02-22T22:21:12.784623Z  185.47.148.110:5060 ---> 0.0.0.0:45695

SIP/2.0 200 OK
Via: SIP/2.0/TCP [our domain]:5671;received=34.151.108.130;branch=z9hG4bK7e73.42822642.0;i=5ca13bc2
Record-Route: <sip:185.47.148.110:5060;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Record-Route: <sip:185.47.148.110;transport=tcp;lr;r2=on;ftag=cfb4f381-ec33-42fa-bd80-366653bccaa4>
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9333 BYE
Content-Length:  0


proto:TCP 2024-02-22T22:21:12.784873Z  0.0.0.0:5165 ---> 10.150.0.2:34717

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj3dcddde1-9306-489f-8a54-55dd74ebf888;alias
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9333 BYE
Content-Length:  0


proto:TCP 2024-02-22T22:21:12.785Z  10.150.0.32:5165 ---> 10.150.0.2:34717

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.150.0.2:6001;received=10.150.0.2;rport=34717;branch=z9hG4bKPj3dcddde1-9306-489f-8a54-55dd74ebf888;alias
Call-ID: 0734a63e-8831-4c93-9b27-8424604e4147
From: <sip:+[from number]@10.150.0.2>;tag=cfb4f381-ec33-42fa-bd80-366653bccaa4
To: <sip:+[dialled number]@10.150.0.32>;tag=188cb894-aff1-4bfd-aed4-f93d8e86f238
CSeq: 9333 BYE
Content-Length:  0
10

For the Asterisk logs, I first ran rtp set debug on; core set verbose 5; core set debug 5.

I did collapse the part that is looped into just one occurrence (the constant cycle of switching codecs during the call), and also removed ast_ari_invoke and calc_rxstamp_and_jitter debug lines that were very noisy and lines with the IP address of the WebRTC client leg of the call (that had no problem).

Key points:

  • Line 263: We originate a channel to our provider
  • Line 594: Handle 200 response from provider (PCMA is the first listed codec as per the SIP trace)
  • Line 711: The first RTP packet we send to our provider is type 09 (G722)

https://pastebin.com/raw/TmTaA92a

11

What is the current complete endpoint configuration?

12

It was as follows during the above test:

[global]
use_callerid_contact=yes

[transport]
type=transport
protocol=tcp
bind=0.0.0.0:6001
external_media_address=[redacted]

[webrtc_client]
type=aor
max_contacts=100000

[webrtc_client]
type=endpoint
allow=!all,ulaw,alaw,h264
aors=webrtc_client
context=webrtc-client
dtls_auto_generate_cert=yes
identify_by=header
max_audio_streams=16
max_video_streams=16
rtp_symmetric=yes
rtp_timeout=30
rtp_timeout_hold=30
webrtc=yes
rewrite_contact=yes
force_rport=yes

[webrtc_client]
type=identify
endpoint=webrtc_client
match_header=X-Invite-Source: app

[voxbone]
type=endpoint
allow=!all,g722,ulaw,alaw
aors=voxbone
context=inbound
direct_media=no
identify_by=header
outbound_auth=voxbone
outgoing_call_offer_pref=local_merge
transport=transport
rtp_timeout=30
rtp_timeout_hold=30

[voxbone]
type=identify
endpoint=voxbone
match_header=X-Invite-Source: voxbone

[voxbone]
type=aor
contact=sip:10.150.0.32:5165

[voxbone]
type=auth
auth_type=md5
md5_cred=[redacted]
realm=voxbone.com
username=[redacted]
13

And if you remove the outgoing_call_offer_pref option?

14

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