Softphone.Pro log file analysis: one-way audio, inbound and outbound calls can't get through

Softphone.Pro users may encounter various cases, such as:

• Agent cannot accept inbound call;
• Agent cannot make outbound call;
• Agent's softphone fails to register on PBX;
• One-way audio;
• Call quality issues.

These situations always occur at inconvenient times and brook no delay. Troubleshooting usually requires the unvolvement of the other business units specialists, admins, PBX support etc. They all need complete information about the issue with examples.

This article shows how to gather and analyze relevant info using Softphone.Pro logs.

Where to find Softphone.Pro log files

Please read this article to learn how to enable logging and set log level. Debug level is needed for the following article to be relevant.

Open app settings, Main tab. In the Application Data section click Browse:

For a custom log path, open the folder specified in Log file path or click Open log file button next to it.

Locate log files in the opened folder (named SoftphoneProLogFile) and open them in any text editor.

For the following examples we use Notepad++.

Finding relevant lines in Notepad++

Notepad++ has advanced search features, including regex. To enable regex search, open SoftphoneProLogFile.txt in Notepad++, go to Search - Find (Ctrl-F), and select Regular expression under Search mode:

Examples include regex patterns to find needed log entries.

Common log acronyms and actions

Frequent log entries

You'll see repeating abbreviations in logs. Here are the most useful ones for diagnostics.

Log entry	Description
RX	Data received from PBX (e.g., INVITE request for incoming call, SIP/2.0 200 OK response to BYE request, received voice packets in call stats). Example: RX 563 bytes Response msg 100/INVITE
TX	Data sent to PBX (e.g., INVITE request for outgoing call, SIP/2.0 180 Ringing response to incoming INVITE request, sent voice packets in call stats). Example: TX 319 bytes Response msg 200/BYE
Making call with acc {ACCOUNT_ID} to <{NUMBER} @{SIP_SERVER}>	Outgoing call start. Shows SIP account ID {ACCOUNT_ID} (0-based index), called number {NUMBER}, and PBX SIP server {SIP_SERVER} to send the message to. Example: Making call with acc #0 to <sip:15551234567@sip.example.com>
TX ... Request	Request sent to PBX from the app (e.g., REGISTER request for SIP account registration or INVITE request for outgoing call). Example: TX 986 bytes Request msg INVITE
RX ... Request	Request received from PBX by the app (e.g., INVITE request for incoming call or BYE request to end call). Example: RX 455 bytes Request msg BYE
TX ... Response	Response sent to PBX from the app (e.g., SIP/2.0 200 OK response to accept incoming call). Example: TX 872 bytes Response msg 200/INVITE
RX ... Response	Response received from PBX by the app (e.g., SIP/2.0 200 OK response to REGISTER request). Example: RX 883 bytes Response msg 200/INVITE
Call-ID: {CALL_ID}	Unique call ID. Usually remains unchanged throughout the call, helping track all related events. Example: Call-ID: e0024d12211e48b0bc6a813d50d0b876
Statistics for ...	Call statistics. Key parameters are explained later. Example: see example 1, example 2.
Incoming call notification window appeared	Incoming call started with notification in the center of the screen.
Small incoming call notification window appeared	Incoming call started with bottom-right notification.
User clicked Answer button	Agent clicked Answer in incoming call or Active Calls window.
User clicked Hangup button	Agent clicked End Call in incoming call or Active Calls window.

Key call statistics to check

Important diagnostic fields to check when analyzing issues.

Stat section	Log entry	Description
RX	Total number of packets	Voice packets received from PBX.
	Total number of discarded packets	Rejected voice packets from PBX.
	Total number of packets lost	Lost voice packets from PBX.
TX	Total number of packets	Voice packets sent to PBX.
	Total number of discarded packets	Rejected voice packets to PBX.
	Total number of packets lost	Lost voice packets to PBX.
Jitter statistic	Average delay	Average voice packet delay (ms).
	Maximum delay	Max voice packet delay (ms).
	Number of lost frames	Lost voice frames.
	Number of discarded frames	Rejected voice frames.

Successful call examples in softphone logs

To find a call in logs and understand how it went, you need:

• Call start time;
• Call end time;
• Number A (caller);
• Number B (callee).

With this info, you can find the call in logs. Below are successful call examples showing what to look for. For SIP details, see RFC 3261. Service data is replaced with [...] for readability.

Find outgoing calls in logs

Outgoing calls start with Making call with acc#0 to <sip:15551234567@sip.example.com>. Find it in Notepad++ with regex Making call with.*to:

[2025-02-19 17:08:27.032 +05:00] [...] !Making call with acc #0 to <sip:15551234567@sip.example.com>

The softphone sends INVITE to start the call. Find with regex TX.*bytes Request msg INVITE:

[2025-02-19 17:08:27.043 +05:00] [...] TX 986 bytes Request msg INVITE/cseq=12879 (tdta1E62D12C) to UDP sip.example.com:5060:
INVITE sip:15551234567@sip.example.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:53457;rport;branch=z9hG4bKPjaac6cbf085e343c98a45c98888a582b1
Max-Forwards: 70
From: "46" <sip:46@sip.example.com>;tag=86bbcc03704a4c08abffe6421c2e13d9
To: <sip:15551234567@sip.example.com>
Contact: "46" <sip:46@192.168.0.141:53457;ob>
Call-ID: e0024d12211e48b0bc6a813d50d0b876
CSeq: 12879 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: SoftphonePro 5.10.0.0
Content-Type: application/sdp
Content-Length:   344

You might see multiple INVITEs if PBX requires authorization (most do). If successful, PBX responds with SIP/2.0 100 Trying and SIP/2.0 180 Ringing, indicating dialing. Find with regex RX.*bytes Response msg:

[2025-02-19 17:08:27.046 +05:00] [...] RX 563 bytes Response msg 100/INVITE/cseq=12880 (rdata1A089814) from UDP sip.example.com:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.141:53457;branch=z9hG4bKPj254d35451f414654aab2a2e0d31bbe81;received=192.168.0.141;rport=53457
From: "46" <sip:46@sip.example.com>;tag=86bbcc03704a4c08abffe6421c2e13d9
To: <sip:15551234567@sip.example.com>
Call-ID: e0024d12211e48b0bc6a813d50d0b876
CSeq: 12880 INVITE
Server: Asterisk PBX 16.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:15551234567@sip.example.com:5060>
Content-Length: 0

...

[2025-02-19 17:08:30.783 +05:00] [...] RX 579 bytes Response msg 180/INVITE/cseq=12880 (rdata1A089814) from UDP sip.example.com:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.141:53457;branch=z9hG4bKPj254d35451f414654aab2a2e0d31bbe81;received=192.168.0.141;rport=53457
From: "46" <sip:46@sip.example.com>;tag=86bbcc03704a4c08abffe6421c2e13d9
To: <sip:15551234567@sip.example.com>;tag=as797f5cc4
Call-ID: e0024d12211e48b0bc6a813d50d0b876
CSeq: 12880 INVITE
Server: Asterisk PBX 16.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:15551234567@sip.example.com:5060>
Content-Length: 0

When callee answers, PBX sends SIP/2.0 200 OK (find with regex RX.*bytes Response msg 200/INVITE):

[2025-02-19 17:08:36.049 +05:00] [...] RX 883 bytes Response msg 200/INVITE/cseq=12880 (rdata1A089814) from UDP sip.example.com:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.141:53457;branch=z9hG4bKPj254d35451f414654aab2a2e0d31bbe81;received=192.168.0.141;rport=53457
From: "46" <sip:46@sip.example.com>;tag=86bbcc03704a4c08abffe6421c2e13d9
To: <sip:15551234567@sip.example.com>;tag=as797f5cc4
Call-ID: e0024d12211e48b0bc6a813d50d0b876
CSeq: 12880 INVITE
Server: Asterisk PBX 16.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:15551234567@sip.example.com:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 260

The softphone confirms with ACK (find with regex TX.*bytes Request msg ACK):

[2025-02-19 17:08:36.147 +05:00] [...] TX 341 bytes Request msg ACK/cseq=18858 (tdta1C6310AC) to UDP sip.example.com:5060:
ACK sip:49@sip.example.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:59229;rport;branch=z9hG4bKPj62e6b035b0694d67947c391e0dba90ba
Max-Forwards: 70
From: "46" <sip:46@sip.example.com>;tag=a2b61c99043e4a688f9cbb499deaa612
To: <sip:15551234567@sip.example.com>;tag=as56c925c0
Call-ID: e0024d12211e48b0bc6a813d50d0b876
CSeq: 12880 ACK
Content-Length:  0

Call started.

If call ends remotely, softphone receives BYE (find with regex RX.*bytes Request msg BYE):

[2025-02-19 17:08:56.630 +05:00] [...] RX 594 bytes Request msg BYE/cseq=102 (rdata1A089814) from UDP sip.example.com:5060:
BYE sip:46@192.168.0.141:53457;ob SIP/2.0
Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK349ee4d8;rport
Max-Forwards: 70
From: <sip:15551234567@sip.example.com>;tag=as797f5cc4
To: "46" <sip:46@sip.example.com>;tag=86bbcc03704a4c08abffe6421c2e13d9
Call-ID: e0024d12211e48b0bc6a813d50d0b876
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.22.0
Proxy-Authorization: Digest username="46", realm="snowwhite", algorithm=MD5, uri="sip:sip.example.com", nonce="7fd922db", response="37fcb37c39b95a8b9959950504fcb71a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Softphone responds with SIP/2.0 200 OK and ends call.

Call stats appear after Media stream destroyed for call:

[2025-02-19 17:08:56.631 +05:00] [...] Media stream destroyed for call with id 5 - get statistic:

Statistics for 20 seconds of the last media stream (call 15551234567 in 2025-02-19 17:08:36 UUID e75b1270-9ae4-44e0-8431-3e1de292dcfb CALL-ID e0024d12211e48b0bc6a813d50d0b876):
RX:
Total number of packets: 1016
Total number of payload/bytes: 162560
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 0
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

TX:
Total number of packets: 1029
Total number of payload/bytes: 164640
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 0
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

Jitter Statistic:
Average delay, in ms: 58
Minimum delay, in ms: 20
Maximum delay, in ms: 80
Standard deviation of delay, in ms: 13
Average burst, in frames: 3
Number of lost frames: 0
Number of discarded frames: 0
Number of empty on GET events: 3

Jitter Settings:
Individual frame size, in bytes: 80
Minimum allowed prefetch, in frms: 1
Maximum allowed prefetch, in frms: 40

Jitter Status:
Current burst level, in frames: 4
Current prefetch value, in frames: 0
Current buffer size, in frames: 1

Codec name: PCMA/8000
Outgoing/incoming codec payload type: 8/8

Non-zero RX: Total number of packets: means voice packets were received from PBX. Non-zero TX: Total number of packets: means voice packets were sent to PBX.

See table for metric details.

Find incoming calls in logs

Incoming calls start with INVITE from PBX. Find with regex RX.*bytes Request msg INVITE:

[2025-02-20 11:29:47.964 +05:00] [...] RX 851 bytes Request msg INVITE/cseq=102 (rdata18A96EF4) from UDP sip.example.com:5060:
INVITE sip:46@192.168.0.141:57753;ob SIP/2.0
Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK125070e4
Max-Forwards: 70
From: "Sales" <sip:+15551234567@sip.example.com>;tag=as3609d76b
To: <sip:46@192.168.0.141:57753;ob>
Contact: <sip:+15551234567@sip.example.com:5060>
Call-ID: 71d507c61a4d08455681617574be07c4@sip.example.com:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.22.0
Date: Thu, 20 Feb 2025 06:29:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

Softphone responds with SIP/2.0 100 Trying and SIP/2.0 180 Ringing. Find with regex TX.*bytes Response msg:

[2025-02-20 11:29:47.975 +05:00] [...] TX 291 bytes Response msg 100/INVITE/cseq=102 (tdta18AF36EC) to UDP sip.example.com:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP sip.example.com:5060;received=sip.example.com;branch=z9hG4bK125070e4
Call-ID: 71d507c61a4d08455681617574be07c4@sip.example.com:5060
From: "Sales" <sip:+15551234567@sip.example.com>;tag=as3609d76b
To: 
CSeq: 102 INVITE
Content-Length:  0

...

[2025-02-20 11:29:47.975 +05:00] [...] TX 474 bytes Response msg 180/INVITE/cseq=102 (tdta18AF46F4) to UDP sip.example.com:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP sip.example.com:5060;received=sip.example.com;branch=z9hG4bK125070e4
Call-ID: 71d507c61a4d08455681617574be07c4@sip.example.com:5060
From: "Sales" <sip:+15551234567@sip.example.com>;tag=as3609d76b
To: <sip:46@192.168.0.141;ob>;tag=425417b023da4d058fa5fd904ec0dde1
CSeq: 102 INVITE
Contact: "46" <sip:46@192.168.0.141:57753;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0

Softphone shows incoming call window (Incoming call notification window appeared for center of the screen, Small incoming call notification window appeared for bottom right corner notification):

[2025-02-20 11:29:48.142 +05:00] [...] Incoming call notification window appeared

Log entry appears when Agent clicks Answer button (User clicked Answer button):

[2025-02-20 11:29:49.786 +05:00] [...] Incoming window: User clicked Answer button

Softphone sends SIP/2.0 200 OK (find with regex TX.*bytes Response msg 200/INVITE):

[2025-02-20 11:29:49.788 +05:00] [...] TX 872 bytes Response msg 200/INVITE/cseq=102 (tdta1D511354) to UDP sip.example.com:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP sip.example.com:5060;received=sip.example.com;branch=z9hG4bK125070e4
Call-ID: 71d507c61a4d08455681617574be07c4@sip.example.com:5060
From: "Sales" <sip:+7991234567@sip.example.com>;tag=as3609d76b
To: <sip:46@192.168.0.141;ob>;tag=425417b023da4d058fa5fd904ec0dde1
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "46" <sip:46@192.168.0.141:57753;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   321

PBX confirms with ACK (find with regex RX.*bytes Request msg ACK):

[2025-02-20 11:29:49.789 +05:00] [...] RX 430 bytes Request msg ACK/cseq=102 (rdata0A25F0FC) from UDP sip.example.com:5060:
ACK sip:46@192.168.0.141:57753;ob SIP/2.0
Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK326221be
Max-Forwards: 70
From: "Sales" <sip:+15551234567@sip.example.com>;tag=as3609d76b
To: <sip:46@192.168.0.141:57753;ob>;tag=425417b023da4d058fa5fd904ec0dde1
Contact: <sip:+15551234567@sip.example.com:5060>
Call-ID: 71d507c61a4d08455681617574be07c4@sip.example.com:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.22.0
Content-Length: 0

Call started.

If Agent ends call another log entry appears (User clicked Hangup button):

[2025-02-20 11:32:51.091 +05:00] [...] Floating window: User clicked Hangup button

Softphone sends BYE (find with regex TX.*bytes Request msg BYE):

[2025-02-20 11:32:51.093 +05:00] [...] TX 421 bytes Request msg BYE/cseq=24355 (tdta1D50B324) to UDP sip.example.com:5060:
BYE sip:+15551234567@sip.example.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.141:57753;rport;branch=z9hG4bKPj4c294bedd77f41198a56296a340f0de6
Max-Forwards: 70
From: <sip:46@192.168.0.141;ob>;tag=425417b023da4d058fa5fd904ec0dde1
To: "Sales" <sip:+15551234567@sip.example.com>;tag=as3609d76b
Call-ID: 71d507c61a4d08455681617574be07c4@sip.example.com:5060
CSeq: 24355 BYE
User-Agent: SoftphonePro 5.10.0.0
Content-Length:  0

Call stats appear after Media stream destroyed for call:

[2025-02-20 11:32:51.093 +05:00] [...] Media stream destroyed for call with id 0 - get statistic:
 
Statistics for 182 seconds of the last media stream (call +15551234567 in 2025-02-20 11:29:49 UUID 72e6fe7d-1f18-4a54-b59b-63fa7f7ef3dc CALL-ID 71d507c61a4d08455681617574be07c4@sip.example.com:5060):
RX:
Total number of packets: 9056
Total number of payload/bytes: 1448960
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 0
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

TX:
Total number of packets: 9067
Total number of payload/bytes: 1450720
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 0
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

Jitter Statistic:
Average delay, in ms: 28
Minimum delay, in ms: 20
Maximum delay, in ms: 80
Standard deviation of delay, in ms: 12
Average burst, in frames: 2
Number of lost frames: 0
Number of discarded frames: 0
Number of empty on GET events: 13

Jitter Settings:
Individual frame size, in bytes: 80
Minimum allowed prefetch, in frms: 1
Maximum allowed prefetch, in frms: 40

Jitter Status:
Current burst level, in frames: 3
Current prefetch value, in frames: 0
Current buffer size, in frames: 0

Non-zero RX: Total number of packets: means voice packets were received from PBX. Non-zero TX: Total number of packets: means voice packets were sent to PBX.

See table for metric details.

Troubleshooting with softphone logs

Common issues are listed with answers to these questions:

1. What happened;
2. What logs show;
3. How to fix.

Full SIP dumps are omitted for brevity.

Agent cannot answer incoming call

What happened: Agent gets incoming call, clicks Answer button, but nothing happens - call doesn't start. Call drops automatically after timeout.

Logs show: Softphone.Pro sends SIP/2.0 200 OK responses to INVITE request but gets no ACK messages from PBX. Many SIP/2.0 200 OK or INVITE events may appear, indicating PBX isn't receiving responses from the app. PBX sends CANCEL to end call.

Example with regex TX .*bytes Response msg 200/INVITE showing multiple responses:

[2025-02-20 11:29:49.788 +05:00] [SoftphonePro] [...] TX 872 bytes Response msg 200/INVITE/cseq=102 (tdta1D511354) to UDP sip.example.com:5060:
SIP/2.0 200 OK
...
[2025-02-20 11:29:50.788 +05:00] [SoftphonePro] [...] TX 872 bytes Response msg 200/INVITE/cseq=102 (tdta1D511354) to UDP sip.example.com:5060:
SIP/2.0 200 OK
...
[2025-02-20 11:29:51.788 +05:00] [SoftphonePro] [...] TX 872 bytes Response msg 200/INVITE/cseq=102 (tdta1D511354) to UDP sip.example.com:5060:
SIP/2.0 200 OK
...
[2025-02-20 11:29:52.788 +05:00] [SoftphonePro] [...] TX 872 bytes Response msg 200/INVITE/cseq=102 (tdta1D511354) to UDP sip.example.com:5060:
SIP/2.0 200 OK

Regex RX.*bytes Request msg ACK returns no results.

PBX sends CANCEL after timeout (find with regex RX.*bytes Request msg CANCEL):

[2025-02-20 11:29:54.132 +05:00] [...] RX 377 bytes Request msg CANCEL/cseq=102 (rdata1B307D4C) from UDP sip.example.com:5060:
CANCEL sip:46@192.168.0.141:1688;ob SIP/2.0
...

Solution: Contact PBX support to check if requests from the app reached PBX. Network issues may also cause this - contact your sysadmin, provider, or PBX support.

Softphone fails to register on PBX with error 408

What happened: Softphone fails to register, shows 408 error. Agent can't receive or make calls.

Logs show: Softphone sends REGISTER requests but gets no SIP/2.0 200 OK responses.

Regex TX.*bytes Request msg REGISTER shows multiple registration attempts:

[2025-02-20 11:39:49.688 +05:00] [...] TX 768 bytes Request msg REGISTER/cseq=39382 (tdta3060592C) to UDP sip.example.com:5060:
REGISTER sip:sip.example.com SIP/2.0
...
[2025-02-20 11:39:54.688 +05:00] [...] TX 768 bytes Request msg REGISTER/cseq=39382 (tdta3060592C) to UDP sip.example.com:5060:
REGISTER sip:sip.example.com SIP/2.0
...
[2025-02-20 11:39:59.688 +05:00] [...] TX 768 bytes Request msg REGISTER/cseq=39382 (tdta3060592C) to UDP sip.example.com:5060:
REGISTER sip:sip.example.com SIP/2.0
...
[2025-02-20 11:40:04.688 +05:00] [...] TX 768 bytes Request msg REGISTER/cseq=39382 (tdta3060592C) to UDP sip.example.com:5060:
REGISTER sip:sip.example.com SIP/2.0

Regex RX.*bytes Response msg 200/REGISTER returns no results.

Solution: Contact PBX support to check if REGISTER requests arrive and why there's no response. Network issues may also cause this - contact your sysadmin, provider, or PBX support.

Agent cannot make outgoing call - no dial tone

What happened: Agent tries to call, call starts without a dial tone. Call doesn't reach the client.

Logs show: Softphone sends INVITE requests but gets no response.

Regex TX.*bytes Request msg INVITE shows multiple call attempts:

[2025-02-19 17:08:27.043 +05:00] [...] TX 986 bytes Request msg INVITE/cseq=12879 (tdta1E62D12C) to UDP sip.example.com:5060:
INVITE sip:15551234567@sip.example.com SIP/2.0
...
[2025-02-19 17:08:28.043 +05:00] [...] TX 986 bytes Request msg INVITE/cseq=12879 (tdta1E62D12C) to UDP sip.example.com:5060:
INVITE sip:15551234567@sip.example.com SIP/2.0
...
[2025-02-19 17:08:29.043 +05:00] [...] TX 986 bytes Request msg INVITE/cseq=12879 (tdta1E62D12C) to UDP sip.example.com:5060:
INVITE sip:15551234567@sip.example.com SIP/2.0
...
[2025-02-19 17:08:30.043 +05:00] [...] TX 986 bytes Request msg INVITE/cseq=12879 (tdta1E62D12C) to UDP sip.example.com:5060:
INVITE sip:15551234567@sip.example.com SIP/2.0

No results for regex RX.*bytes Response msg with SIP/2.0 100 Trying, SIP/2.0 180 Ringing or SIP/2.0 200 OK responses.

Solution: Contact PBX support to check if INVITE requests arrive and why there's no response. Network issues may also cause this - contact your sysadmin, provider, or PBX support.

Agent doesn't hear client or vice versa (one-way audio)

What happened: Agent makes/receives call. Agent hears client but client doesn't hear Agent (or vice versa).

Logs show: In call stats (find with regex Statistics for.*seconds of the last media stream), Total number of packets: is 0. If 0 in RX, voice packets didn't reach softphone (Agent doesn't hear client):

[2025-02-20 11:32:51.093 +05:00] [...] Media stream destroyed for call with id 0 - get statistic:

Statistics for 182 seconds of the last media stream (call +15551234567 in 2025-02-20 11:29:49 UUID 72e6fe7d-1f18-4a54-b59b-63fa7f7ef3dc CALL-ID 71d507c61a4d08455681617574be07c4@sip.example.com:5060):
RX:
Total number of packets: 0
Total number of payload/bytes: 0
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 0
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

TX:
Total number of packets: 9067
Total number of payload/bytes: 1450720
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 0
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0
...

If 0 in TX, voice packets didn't reach PBX (client doesn't hear Agent).

Solution: If using antivirus/firewall, try disabling it. If it is fixed, add softphone to exceptions (example for Kaspersky). In SIP settings, enable Allow IP rewrite setting and change Firewall traversal method (if app is configured from Team Dashboard use softphone provisioning). Network issues may also cause this - contact your sysadmin, provider, or PBX support.

Call quality issues - choppy audio

What happened: Agent makes/receives call. Both hear each other but audio is choppy.

Logs show: In call stats (find with regex Statistics for.*seconds of the last media stream), Total number of packets lost: is non-zero. If >0 in RX, voice packets from PBX are lost (Agent hears client poorly). If >0 in TX, voice packets to PBX are lost (client hears Agent poorly).

If no packet loss (0 in Total number of packets lost:), check Number of lost frames and Number of discarded frames in Jitter Statistic.

If >0 here, connection issues exist (high load, low bandwidth).

Example with both packet loss and connection issues:

[2025-02-20 11:42:51.093 +05:00] [...] Media stream destroyed for call with id 0 - get statistic:
  
Statistics for 213 seconds of the last media stream (call +15551234567 in 2025-02-20 11:29:49 UUID 72e6fe7d-1f18-4a54-b59b-63fa7f7ef3dc CALL-ID 71d507c61a4d08455681617574be07c4@sip.example.com:5060):
RX:
Total number of packets: 9056
Total number of payload/bytes: 1448960
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 123
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

TX:
Total number of packets: 9067
Total number of payload/bytes: 1450720
Total number of discarded packets: 0
Total number of out of order packets: 0
Total number of packets lost: 456
Total number of duplicates packets: 0
Burst/sequential packet lost detected: 0
Random packet lost detected: 0

Jitter Statistic:
Average delay, in ms: 28
Minimum delay, in ms: 20
Maximum delay, in ms: 80
Standard deviation of delay, in ms: 12
Average burst, in frames: 2
Number of lost frames: 567
Number of discarded frames: 702
Number of empty on GET events: 13

Jitter Settings:
Individual frame size, in bytes: 80
Minimum allowed prefetch, in frms: 1
Maximum allowed prefetch, in frms: 40

Jitter Status:
Current burst level, in frames: 3
Current prefetch value, in frames: 0
Current buffer size, in frames: 0

Solution: try to disable antivirus/firewall. If it helps, add the softphone to exceptions (example for Kaspersky Antivirus). Should you use Wi-Fi, try wired connection. The issue can be caused by network issues or high load so contact your system admin, provider, or PBX support.

See all articles