No sound and no hangup event one way, all good the other way

Asterisk Asterisk Support
1

I’m using the latest Asterisk (22.2).
On one side I have a SIP trunk provider which allows calls to regular phones.
On the other side I have a SIP phone (we use MicroSIP softphone) and WebRTC (website).
We only have outgoing calls, from softphone\website to regular phones.
Calls go through and on a phone I can hear sound coming from the softphone or website.
But, there’s no sound back from the phone on the softphone\website.
Also the softphone\website do not receive the hangup event, when a user stops the call on the phone.
MixMonitor on the Asterisk server records the sound coming from the phone, it confirms the sound reaches Asterisk.
When a phone user hangs up, a Bye request is sent to Asterisk and Asterisk responds with 200 OK.
(BTW, I do not see that Asterisk sends the similar Bye event to the softphone. or maybe it should not be displayed in the logs?)
So, it seems the connection between Asterisk and the provider\phones is OK, sound works, hangups work.
But the connection between Asterisk and the softphone\webrtc work one way only.
The server with Asterisk is directly connected to the internet. No NAT.
The softphone and computers that access the website are directly connected to the internet.
There are some firewall rules on the Asterisk server, but disabling the firewall does not change anything, calls go through with one-way sound.
Please advise, what could be wrong?

Here’s a part of pjsip.conf related to softphone\webrtc:
[general]
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/certificate/asterisk.pem
tlscafile=/etc/asterisk/certificate/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1_2
tlsbindaddr=193.169.XXX.XXX:8089
sipdebug=yes

rtpstart=10000
rtpend=65535

[udp-transport]
type=transport
protocol=udp
bind=0.0.0.0

; MicroSIP
[1001];
type=endpoint
context=outgoing
disallow=all
allow=ulaw
transport=udp-transport
auth=1001
aors=1001

[1001]
type=auth
auth_type=userpass
username=1001
password=xxxxxx

[1001]
type = aor
max_contacts = 1

; WebRTC
[1060]
type=aor
max_contacts=1
remove_existing=yes

[1060-auth]
type=auth
auth_type=userpass
username=1060
password=xxxxxx

[1060]
type=endpoint
aors=1060
auth=1060-auth
transport=udp-transport
context=outgoing
webrtc=yes
disallow=all
allow=ulaw

2

These don’t exist in PJSIP here. Did you copy them from elsewhere such as chan_sip?

This is almost always network related in some way. You need to show logs.

3

I copied these [general] lines from some guide. If they aren’t relevant, will try to remove them.
What would be the best format of logs? Which portion? I look at debug.log and even a single short call is extensive (like, 32kb)

4

A complete call.

From /var/log/full, which needs to be enabled, and with verbosity at least 5, and “pjsip set logger on” in effect. Don’t enable debug logging, unless asked for it, and if asked for it, try to edit out irrelevant detail. Debug logging is generally too noisy for initial investigations.

5

Here is full.log with error,warning,verbose(5) for a 5 second call from MicroSIP to mobile phone, hang up on the phone:
[Mar 20 01:14:57] VERBOSE[101133][C-00000001] pbx.c: Executing [380685978789@outgoing:1] MixMonitor(“PJSIP/1001-00000000”, “/home/ubuntu/20250320-011457_1001_380685978789.wav”) in new stack
[Mar 20 01:14:57] VERBOSE[101133][C-00000001] pbx.c: Executing [380685978789@outgoing:2] Dial(“PJSIP/1001-00000000”, “PJSIP/380685978789@380630515371”) in new stack
[Mar 20 01:14:57] VERBOSE[101134][C-00000001] app_mixmonitor.c: Begin MixMonitor Recording PJSIP/1001-00000000
[Mar 20 01:14:57] VERBOSE[101133][C-00000001] app_dial.c: Called PJSIP/380685978789@380630515371
[Mar 20 01:15:01] VERBOSE[101084] res_rtp_asterisk.c: 0x7576980763a0 – Strict RTP learning after remote address set to: 212.58.160.197:58532
[Mar 20 01:15:01] VERBOSE[101133][C-00000001] app_dial.c: PJSIP/380630515371-00000001 is making progress passing it to PJSIP/1001-00000000
[Mar 20 01:15:01] VERBOSE[101084] res_rtp_asterisk.c: 0x757698061f70 – Strict RTP learning after remote address set to: 192.168.50.217:4016
[Mar 20 01:15:01] VERBOSE[101084] res_rtp_asterisk.c: 0x757698061f70 – Strict RTP learning after remote address set to: 192.168.50.217:4016
[Mar 20 01:15:01] VERBOSE[101133][C-00000001] app_dial.c: PJSIP/380630515371-00000001 is making progress passing it to PJSIP/1001-00000000
[Mar 20 01:15:01] VERBOSE[101084] res_rtp_asterisk.c: 0x757698061f70 – Strict RTP learning after remote address set to: 192.168.50.217:4016
[Mar 20 01:15:01] VERBOSE[101133][C-00000001] app_dial.c: PJSIP/380630515371-00000001 is ringing
[Mar 20 01:15:01] VERBOSE[101084] res_rtp_asterisk.c: 0x757698061f70 – Strict RTP learning after remote address set to: 192.168.50.217:4016
[Mar 20 01:15:01] VERBOSE[101133][C-00000001] res_rtp_asterisk.c: 0x7576980763a0 – Strict RTP switching to RTP target address 212.58.160.197:58532 as source
[Mar 20 01:15:01] VERBOSE[101133][C-00000001] res_rtp_asterisk.c: (0x75769804ca30) RTP audio difference is 314589176 set mark
[Mar 20 01:15:03] VERBOSE[101133][C-00000001] app_dial.c: PJSIP/380630515371-00000001 answered PJSIP/1001-00000000
[Mar 20 01:15:03] VERBOSE[101150][C-00000001] bridge_channel.c: Channel PJSIP/380630515371-00000001 joined ‘simple_bridge’ basic-bridge <3f09abc2-6194-4e0a-a727-6df6348a1f1d>
[Mar 20 01:15:03] VERBOSE[101133][C-00000001] bridge_channel.c: Channel PJSIP/1001-00000000 joined ‘simple_bridge’ basic-bridge <3f09abc2-6194-4e0a-a727-6df6348a1f1d>
[Mar 20 01:15:03] VERBOSE[101084] res_rtp_asterisk.c: 0x757698061f70 – Strict RTP learning after remote address set to: 192.168.50.217:4016
[Mar 20 01:15:06] VERBOSE[101150][C-00000001] res_rtp_asterisk.c: 0x7576980763a0 – Strict RTP learning complete - Locking on source address 212.58.160.197:58532
[Mar 20 01:15:08] VERBOSE[101150][C-00000001] bridge_channel.c: Channel PJSIP/380630515371-00000001 left ‘simple_bridge’ basic-bridge <3f09abc2-6194-4e0a-a727-6df6348a1f1d>
[Mar 20 01:15:08] VERBOSE[101133][C-00000001] bridge_channel.c: Channel PJSIP/1001-00000000 left ‘simple_bridge’ basic-bridge <3f09abc2-6194-4e0a-a727-6df6348a1f1d>
[Mar 20 01:15:08] VERBOSE[101133][C-00000001] pbx.c: Spawn extension (outgoing, 380685978789, 2) exited non-zero on ‘PJSIP/1001-00000000’
[Mar 20 01:15:08] VERBOSE[101134][C-00000001] app_mixmonitor.c: MixMonitor close filestream (mixed)
[Mar 20 01:15:08] VERBOSE[101134][C-00000001] app_mixmonitor.c: End MixMonitor Recording PJSIP/1001-00000000

6

“pjsip set logger on” was not in effect.

7

Sorry, I thought it was.
I would’ve attached the new log as a file but new users cannot attach files.
And new users cannot post texts with more than 2 links and probably the address of the pbx server is considered a link .. let me think how to post it.

So, below are two parts of full.log file, the first half was automatically hidden by the local protection system but should appear after review.

8 
[Mar 20 02:02:05] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (1199 bytes) from TLS:62.216.59.42:60835 --->
INVITE sip:380685978789@mypbx.ua;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPjed66b7cb97324dcb902865b11c3eaec2;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>
Contact: "rua" <sip:1001@192.168.50.217:60835;transport=TLS;ob>
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28959 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: MicroSIP/3.21.6
Content-Type: application/sdp
Content-Length:   507

v=0
o=- 3951424992 3951424992 IN IP4 192.168.50.217
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 96 101 102
c=IN IP4 192.168.50.217
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.50.217
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=ssrc:1361197531 cname:31a61325769779a3

[Mar 20 02:02:05] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (577 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPjed66b7cb97324dcb902865b11c3eaec2;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=z9hG4bKPjed66b7cb97324dcb902865b11c3eaec2
CSeq: 28959 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1742428925/0ec2e38d6e132e6449621b3ba26df9b7",opaque="1c8bb87759ebc323",algorithm=MD5,qop="auth"
Server: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:05] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (423 bytes) from TLS:62.216.59.42:60835 --->
ACK sip:380685978789@mypbx.ua;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPjed66b7cb97324dcb902865b11c3eaec2;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=z9hG4bKPjed66b7cb97324dcb902865b11c3eaec2
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28959 ACK
Content-Length:  0


[Mar 20 02:02:05] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (1516 bytes) from TLS:62.216.59.42:60835 --->
INVITE sip:380685978789@mypbx.ua;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPj76be2311f18c40eb9b6e5934d3f29f9b;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>
Contact: "rua" <sip:1001@192.168.50.217:60835;transport=TLS;ob>
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28960 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: MicroSIP/3.21.6
Authorization: Digest username="1001", realm="asterisk", nonce="1742428925/0ec2e38d6e132e6449621b3ba26df9b7", uri="sip:380685978789@mypbx.ua;transport=tls", response="508f1f07447d34ee8331a7429ded6593", algorithm=MD5, cnonce="724b400a298440b2beefb66d87f02765", opaque="1c8bb87759ebc323", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   507

v=0
o=- 3951424992 3951424992 IN IP4 192.168.50.217
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 96 101 102
c=IN IP4 192.168.50.217
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.50.217
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=ssrc:1361197531 cname:31a61325769779a3

[Mar 20 02:02:05] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (379 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPj76be2311f18c40eb9b6e5934d3f29f9b;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>
CSeq: 28960 INVITE
Server: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:05] VERBOSE[102115][C-00000005] pbx.c: Executing [380685978789@outgoing:1] MixMonitor("PJSIP/1001-00000008", "/home/ubuntu/20250320-020205_1001_380685978789.wav") in new stack
[Mar 20 02:02:05] VERBOSE[102115][C-00000005] pbx.c: Executing [380685978789@outgoing:2] Dial("PJSIP/1001-00000008", "PJSIP/380685978789@380630515371") in new stack
[Mar 20 02:02:05] VERBOSE[102117][C-00000005] app_mixmonitor.c: Begin MixMonitor Recording PJSIP/1001-00000008
[Mar 20 02:02:05] VERBOSE[102115][C-00000005] app_dial.c: Called PJSIP/380685978789@380630515371
[Mar 20 02:02:05] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (1057 bytes) to TLS:212.58.160.196:5081 --->
INVITE sip:380685978789@provider.ua:5081 SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPje8c25487-4878-4ac5-835a-b6c685d878b4;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>
Contact: <sip:380630515371@193.169.240.153:5061;transport=TLS>
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19450 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Content-Type: application/sdp
Content-Length:   326

v=0
o=- 258583860 258583860 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 50742 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y3ZUMdnefQyHwLzxjfv5IbbtCpPDijIJv37vEsMw
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[Mar 20 02:02:05] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (530 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TLS 193.169.240.153:5061;rport=36815;branch=z9hG4bKPje8c25487-4878-4ac5-835a-b6c685d878b4;alias;received=193.169.240.153
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=d3871885b6b7d6ce1ed68cd3736d3b31.4bf79303
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19450 INVITE
Proxy-Authenticate: Digest realm="193.169.240.153", nonce="Z9tcKWfbWv0aqiZQvbL8AXMHirTJWKLx"
Content-Length: 0


[Mar 20 02:02:05] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (463 bytes) to TLS:212.58.160.196:5081 --->
ACK sip:380685978789@provider.ua:5081 SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPje8c25487-4878-4ac5-835a-b6c685d878b4;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=d3871885b6b7d6ce1ed68cd3736d3b31.4bf79303
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19450 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:05] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (1268 bytes) to TLS:212.58.160.196:5081 --->
INVITE sip:380685978789@provider.ua:5081 SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPj034fec3a-2849-4aac-a0eb-48658883f799;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>
Contact: <sip:380630515371@193.169.240.153:5061;transport=TLS>
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Proxy-Authorization: Digest username="380630515371", realm="193.169.240.153", nonce="Z9tcKWfbWv0aqiZQvbL8AXMHirTJWKLx", uri="sip:380685978789@provider.ua:5081", response="5300d0629ff3379fe122d4f62e282784"
Content-Type: application/sdp
Content-Length:   326

v=0
o=- 258583860 258583860 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 50742 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:y3ZUMdnefQyHwLzxjfv5IbbtCpPDijIJv37vEsMw
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

[Mar 20 02:02:05] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (399 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TLS 193.169.240.153:5061;rport=36815;branch=z9hG4bKPj034fec3a-2849-4aac-a0eb-48658883f799;alias;received=193.169.240.153
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 INVITE
Content-Length: 0


[Mar 20 02:02:08] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (953 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 183 Session Progress
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 INVITE
Require: 100rel
RSeq: 500
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Content-Type: application/sdp
Content-Length:  367
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPj034fec3a-2849-4aac-a0eb-48658883f799;alias
Contact: <sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls>

v=0
o=- 3342024 3342024 IN IP4 212.58.160.197
s=-
c=IN IP4 212.58.160.197
t=0 0
m=audio 25588 RTP/SAVP 0 101
c=IN IP4 212.58.160.197
b=RR:0
b=RS:0
a=maxptime:40
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:25589
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qkGKKGpw+IGsMvVdA8eeXGZJYk0mjxZy8QDBMmX1
a=ptime:20

[Mar 20 02:02:08] VERBOSE[102101] res_rtp_asterisk.c: 0x5b195779ad50 -- Strict RTP learning after remote address set to: 212.58.160.197:25588
[Mar 20 02:02:08] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (490 bytes) to TLS:212.58.160.196:5081 --->
PRACK sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPja7928694-5b9b-4e30-b34f-367e8a280654;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19452 PRACK
RAck: 500 19451 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:08] VERBOSE[102115][C-00000005] app_dial.c: PJSIP/380630515371-00000009 is making progress passing it to PJSIP/1001-00000008
[Mar 20 02:02:08] VERBOSE[102101] res_rtp_asterisk.c: 0x5b1957773b90 -- Strict RTP learning after remote address set to: 192.168.50.217:4006
[Mar 20 02:02:08] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (1044 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPj76be2311f18c40eb9b6e5934d3f29f9b;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28960 INVITE
Server: Asterisk PBX 22.2.0
Contact: <sip:193.169.240.153:5061;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   412

v=0
o=- 3951424992 3951424994 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 96 101 102
a=rtpmap:0 PCMU/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:08] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (384 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 200 OK
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19452 PRACK
Content-Length:    0
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPja7928694-5b9b-4e30-b34f-367e8a280654;alias


[Mar 20 02:02:08] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (977 bytes) from TLS:62.216.59.42:60835 --->
UPDATE sip:193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPjfc657f69f00049668ff92d6623a96343;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
Contact: "rua" <sip:1001@192.168.50.217:60835;transport=TLS;ob>
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28961 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   379

v=0
o=- 3951424992 3951424993 IN IP4 192.168.50.217
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101 102
c=IN IP4 192.168.50.217
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.50.217
a=ssrc:1361197531 cname:31a61325769779a3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=sendrecv

[Mar 20 02:02:08] VERBOSE[102101] res_rtp_asterisk.c: 0x5b1957773b90 -- Strict RTP learning after remote address set to: 192.168.50.217:4006
[Mar 20 02:02:08] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (962 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPjfc657f69f00049668ff92d6623a96343;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28961 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:193.169.240.153:5061;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.2.0
Content-Type: application/sdp
Content-Length:   243

v=0
o=- 3951424992 3951424995 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:08] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (504 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 183 Session Progress
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 INVITE
Require: 100rel
RSeq: 501
Content-Length:    0
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPj034fec3a-2849-4aac-a0eb-48658883f799;alias
Contact: <sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls>
9 
[Mar 20 02:02:08] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (490 bytes) to TLS:212.58.160.196:5081 --->
PRACK sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPje97dd8b4-989f-4d06-90e4-fa8f963002d5;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19453 PRACK
RAck: 501 19451 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:08] VERBOSE[102115][C-00000005] app_dial.c: PJSIP/380630515371-00000009 is making progress passing it to PJSIP/1001-00000008
[Mar 20 02:02:08] VERBOSE[102102] res_pjsip_logger.c: <--- Transmitting SIP response (1044 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPj76be2311f18c40eb9b6e5934d3f29f9b;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28960 INVITE
Server: Asterisk PBX 22.2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:193.169.240.153:5061;transport=TLS>
Content-Type: application/sdp
Content-Length:   412

v=0
o=- 3951424992 3951424994 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 96 101 102
a=rtpmap:0 PCMU/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:08] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (384 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 200 OK
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19453 PRACK
Content-Length:    0
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPje97dd8b4-989f-4d06-90e4-fa8f963002d5;alias


[Mar 20 02:02:08] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (991 bytes) from TLS:62.216.59.42:60835 --->
UPDATE sip:193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPjf4adc04cb008400c8fa5dc58d7a90fc1;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
Contact: "rua" <sip:1001@192.168.50.217:60835;transport=TLS;ob>
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28962 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length:   379

v=0
o=- 3951424992 3951424995 IN IP4 192.168.50.217
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101 102
c=IN IP4 192.168.50.217
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.50.217
a=ssrc:1361197531 cname:31a61325769779a3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=sendrecv

[Mar 20 02:02:08] VERBOSE[102101] res_rtp_asterisk.c: 0x5b1957773b90 -- Strict RTP learning after remote address set to: 192.168.50.217:4006
[Mar 20 02:02:08] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (962 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPjf4adc04cb008400c8fa5dc58d7a90fc1;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28962 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:193.169.240.153:5061;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.2.0
Content-Type: application/sdp
Content-Length:   243

v=0
o=- 3951424992 3951424995 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:09] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (495 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 180 Ringing
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 INVITE
Require: 100rel
RSeq: 502
Content-Length:    0
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPj034fec3a-2849-4aac-a0eb-48658883f799;alias
Contact: <sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls>


[Mar 20 02:02:09] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (490 bytes) to TLS:212.58.160.196:5081 --->
PRACK sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPj69fc8003-ad89-4180-ab54-7adadc0d4e4a;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19454 PRACK
RAck: 502 19451 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:09] VERBOSE[102115][C-00000005] app_dial.c: PJSIP/380630515371-00000009 is ringing
[Mar 20 02:02:09] VERBOSE[102102] res_pjsip_logger.c: <--- Transmitting SIP response (1044 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPj76be2311f18c40eb9b6e5934d3f29f9b;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28960 INVITE
Server: Asterisk PBX 22.2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:193.169.240.153:5061;transport=TLS>
Content-Type: application/sdp
Content-Length:   412

v=0
o=- 3951424992 3951424994 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 96 101 102
a=rtpmap:0 PCMU/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:09] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (384 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 200 OK
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19454 PRACK
Content-Length:    0
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPj69fc8003-ad89-4180-ab54-7adadc0d4e4a;alias


[Mar 20 02:02:09] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (991 bytes) from TLS:62.216.59.42:60835 --->
UPDATE sip:193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPje11d92c56d0647f9a8dbdde96242da5b;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
Contact: "rua" <sip:1001@192.168.50.217:60835;transport=TLS;ob>
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28963 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length:   379

v=0
o=- 3951424992 3951424997 IN IP4 192.168.50.217
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101 102
c=IN IP4 192.168.50.217
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.50.217
a=ssrc:1361197531 cname:31a61325769779a3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=sendrecv

[Mar 20 02:02:09] VERBOSE[102101] res_rtp_asterisk.c: 0x5b1957773b90 -- Strict RTP learning after remote address set to: 192.168.50.217:4006
[Mar 20 02:02:09] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (962 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPje11d92c56d0647f9a8dbdde96242da5b;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28963 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:193.169.240.153:5061;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.2.0
Content-Type: application/sdp
Content-Length:   243

v=0
o=- 3951424992 3951424995 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:09] VERBOSE[102115][C-00000005] res_rtp_asterisk.c: 0x5b195779ad50 -- Strict RTP switching to RTP target address 212.58.160.197:25588 as source
[Mar 20 02:02:09] VERBOSE[102115][C-00000005] res_rtp_asterisk.c: (0x5b1957772fe0) RTP audio difference is 339687264 set mark
[Mar 20 02:02:10] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP response (609 bytes) from TLS:212.58.160.196:5081 --->
SIP/2.0 200 OK
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
P-Asserted-Identity: <sip:685978789@KIES4.MSS.LIFECELL.UA;user=phone>
Accept: application/sdp
Content-Length:    0
Via: SIP/2.0/TLS 193.169.240.153:5061;received=193.169.240.153;rport=36815;branch=z9hG4bKPj034fec3a-2849-4aac-a0eb-48658883f799;alias
Contact: <sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls>


[Mar 20 02:02:10] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP request (462 bytes) to TLS:212.58.160.196:5081 --->
ACK sip:atpsh-67d88040-28e593-2572@212.58.160.196:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 193.169.240.153:5061;rport;branch=z9hG4bKPja868f4a8-ddf6-45e8-96a7-b243a40d29d4;alias
From: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
To: <sip:380685978789@provider.ua>;tag=09064080649120
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 19451 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:10] VERBOSE[102115][C-00000005] app_dial.c: PJSIP/380630515371-00000009 answered PJSIP/1001-00000008
[Mar 20 02:02:10] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (1131 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPj76be2311f18c40eb9b6e5934d3f29f9b;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28960 INVITE
Server: Asterisk PBX 22.2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Contact: <sip:193.169.240.153:5061;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   412

v=0
o=- 3951424992 3951424994 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 96 101 102
a=rtpmap:0 PCMU/8000
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:10] VERBOSE[102127][C-00000005] bridge_channel.c: Channel PJSIP/380630515371-00000009 joined 'simple_bridge' basic-bridge <fe3239d6-7946-4966-811f-1d8e3bdf4c66>
[Mar 20 02:02:10] VERBOSE[102115][C-00000005] bridge_channel.c: Channel PJSIP/1001-00000008 joined 'simple_bridge' basic-bridge <fe3239d6-7946-4966-811f-1d8e3bdf4c66>
[Mar 20 02:02:10] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (410 bytes) from TLS:62.216.59.42:60835 --->
ACK sip:193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPj3aa5f0bb480641a3a1838a9ad21786b5;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28960 ACK
Content-Length:  0


[Mar 20 02:02:10] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (991 bytes) from TLS:62.216.59.42:60835 --->
UPDATE sip:193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPj9c509f8409d943b882f8a6c05d861778;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
Contact: "rua" <sip:1001@192.168.50.217:60835;transport=TLS;ob>
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28964 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length:   379

v=0
o=- 3951424992 3951424999 IN IP4 192.168.50.217
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101 102
c=IN IP4 192.168.50.217
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.50.217
a=ssrc:1361197531 cname:31a61325769779a3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/48000
a=fmtp:102 0-16
a=sendrecv

[Mar 20 02:02:10] VERBOSE[102101] res_rtp_asterisk.c: 0x5b1957773b90 -- Strict RTP learning after remote address set to: 192.168.50.217:4006
[Mar 20 02:02:10] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (962 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPj9c509f8409d943b882f8a6c05d861778;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28964 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:193.169.240.153:5061;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.2.0
Content-Type: application/sdp
Content-Length:   243

v=0
o=- 3951424992 3951424995 IN IP4 193.169.240.153
s=Asterisk
c=IN IP4 193.169.240.153
t=0 0
m=audio 54402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

[Mar 20 02:02:13] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (426 bytes) from TLS:212.58.160.196:5081 --->
BYE sip:380630515371@193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 212.58.160.196:5081;branch=z9hG4bKfb6d.f3db00e507e3c36b669bbb592afda604.0
From: <sip:380685978789@provider.ua>;tag=09064080649120
To: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
Max-Forwards: 69
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
CSeq: 1 BYE
Reason: Q.850;cause=16
Content-Length:    0


[Mar 20 02:02:13] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (397 bytes) to TLS:212.58.160.196:5081 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 212.58.160.196:5081;rport=5081;received=212.58.160.196;branch=z9hG4bKfb6d.f3db00e507e3c36b669bbb592afda604.0
Call-ID: 0e5d7899-0420-4b2f-abfe-d1fbc6bad194
From: <sip:380685978789@provider.ua>;tag=09064080649120
To: <sip:380630515371@193.169.240.153>;tag=fd0275a7-9c23-4e79-91d8-61b27350194b
CSeq: 1 BYE
Server: Asterisk PBX 22.2.0
Content-Length:  0


[Mar 20 02:02:13] VERBOSE[102127][C-00000005] bridge_channel.c: Channel PJSIP/380630515371-00000009 left 'simple_bridge' basic-bridge <fe3239d6-7946-4966-811f-1d8e3bdf4c66>
[Mar 20 02:02:13] VERBOSE[102115][C-00000005] bridge_channel.c: Channel PJSIP/1001-00000008 left 'simple_bridge' basic-bridge <fe3239d6-7946-4966-811f-1d8e3bdf4c66>
[Mar 20 02:02:13] VERBOSE[102115][C-00000005] pbx.c: Spawn extension (outgoing, 380685978789, 2) exited non-zero on 'PJSIP/1001-00000008'
[Mar 20 02:02:13] VERBOSE[102117][C-00000005] app_mixmonitor.c: MixMonitor close filestream (mixed)
[Mar 20 02:02:13] VERBOSE[102117][C-00000005] app_mixmonitor.c: End MixMonitor Recording PJSIP/1001-00000008
[Mar 20 02:02:14] VERBOSE[101083] res_pjsip_logger.c: <--- Received SIP request (439 bytes) from TLS:62.216.59.42:60835 --->
BYE sip:193.169.240.153:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.50.217:60835;rport;branch=z9hG4bKPjcbceca8e11674beb9463beebf8df16b8;alias
Max-Forwards: 70
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
Call-ID: bff75511f731487cab264955c89f2c75
CSeq: 28965 BYE
User-Agent: MicroSIP/3.21.6
Content-Length:  0


[Mar 20 02:02:14] VERBOSE[102101] res_pjsip_logger.c: <--- Transmitting SIP response (413 bytes) to TLS:62.216.59.42:60835 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.50.217:60835;rport=60835;received=62.216.59.42;branch=z9hG4bKPjcbceca8e11674beb9463beebf8df16b8;alias
Call-ID: bff75511f731487cab264955c89f2c75
From: "rua" <sip:1001@mypbx.ua>;tag=a33999eb9db64723aa6d88c8860015ec
To: <sip:380685978789@mypbx.ua>;tag=977dcb11-b629-4e8b-80eb-fbf7a79659ae
CSeq: 28965 BYE
Server: Asterisk PBX 22.2.0
Content-Length:  0
10

The incoming request originated behind NAT and doesn’t compensate for that. For media to flow towards the caller, you are going to need rtp-symmetric to be enabled (probably also force-rport and rewrite-contact), and the caller to send media to you. The latter is the only way that Asterisk can learn the true public media address, for the caller.

You are also going to need to ensure that you firwall has static port forwarding and firewall accept rules for Asterisk’s RTP port range, as it will never receive any media which it can use to construct dynamic rules, until Asterisk has learned the correct address, as Asterisk will, provisionally, be sending to the unroueable 192.168/16 network.

Alternatively, if possible, configure the remote endpoint to send its public address in Via, and Contact, headers, and the c= line in its SDP.

11

rtpkeepalive = 1

12

rtpkeepalive doesn’t solve the problem of the wrong address in the c= line. What it does do is a avoid a dynamic rule in the router timing out, when there is a gap in the audio.

13

To clarify, I have these elements,
PC (MicroSIP or Chrome)(192.168/16) - WiFi Router - (internet provider) - Asterisk server (firewall) - SIP provider - Phone

I understand that NAT can prevent calls, but my PC is behind a transparent WiFi router with no firewall, can it really block a part of a call?
I tried to connect my PC through mobile internet (replace the WiFi router with mobile provider), it works the same, no inbound sound. The log is similar to the one I posted.

Also I tried to disable the firewall on the Asterisk server, it had no effect.

I tried to add rtp_symmetric, force-rport, rewrite-contact parameters to the pjsip.conf to the enpoint, no effect:
[1001]
type=endpoint
context=outgoing
disallow=all
allow=ulaw
allow=opus
transport=udp-transport
auth=1001
aors=1001

direct_media=no
rtp_symmetric=yes
force-rport=yes
rewrite-contact=yes

and this as well
rtpkeepalive=1

14

Sorry, I was wrong. Problem resolved. I got sound both ways.
I had ufw on the Asterisk server and it was not disabled.

Thank you for your support!

15

So, I thought it was solved but it’s not.

As david551 recommended I added a setting rtp-symmetric=yes and disabled ufw firewall on the Asterisk server and got sound both ways on the MicroSIP phone.
BTW, the c= line still shows my internal ip behind WiFi router:

res_pjsip_logger.c: <— Received SIP request (990 bytes) from TLS:62.216.59.42:57831 —>
c=IN IP4 192.168.50.217

But MicroSIP still does not get “Bye” events when the other user hangs up. That is issue #1 (though it’s lower priority)

Issue #2 (most important) is that I still do not get inbound sound on WebRTC (and hangups #3).
“rtp-symmetric” did not help there.
I saw another advice: nat=force_rport,comedia - did not help either.

Yes, the logs show different IPs in c= (identical to above for MicroSIP, but there I have sound).

I tried to set stun:null in jsSIP configuration (stun_servers: null), no effect.

I tried the sample config and there’s no sound either (though MixMonitor does record the congrats voice)

exten => 200,1,Ans> wer()
same => n,Playback(demo-congrats)
same => n,Hangup()

(tried both PC via WiFi router and PC via mobile internet, same)

RTP log shows data is sent

[Mar 21 13:01:28] VERBOSE[148817] res_rtp_asterisk.c: 0x7576c4018690 – Strict RTP learning after remote address set to: 192.168.50.217:55107
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: 0x7576c4018690 – Strict RTP learning after ICE completion
[Mar 21 13:01:28] VERBOSE[101097] res_rtp_asterisk.c: 0x7576c4018690 – Strict RTP learning after remote address set to: 62.216.59.42:55107
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: 0x7576c4018690 – Strict RTP switching to RTP target address 62.216.59.42:55107 as source
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:55107 (type 00, seq 012518, ts 3251806296, len 000170)
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] pbx.c: Executing [380685978789@rtc60:3] Playback(“PJSIP/1060-0000006e”, “hello-world”) in new stack
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:55107 (via ICE) (type 00, seq 025103, ts 000160, len 000170)
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] file.c: <PJSIP/1060-0000006e> Playing ‘hello-world.gsm’ (language ‘en’)
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:55107 (type 00, seq 012519, ts 3251806456, len 000170)
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:55107 (via ICE) (type 00, seq 025104, ts 000320, len 000170)
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: Got RTP packet from 62.216.59.42:55107 (type 00, seq 012520, ts 3251806616, len 000170)
[Mar 21 13:01:28] VERBOSE[148921][C-0000003b] res_rtp_asterisk.c: Sent RTP packet to 62.216.59.42:55107 (via ICE) (type 00, seq 025105, ts 000480, len 000170)

Here’s my current pjsip config for WebRTC clients:

[1060]
type=endpoint
aors=1060
auth=1060-auth
context=rtc60
webrtc=yes
disallow=all
allow=ulaw
allow=opus

16

This relates to the, obsolete, chan_sip.

nat=comedia is rtp_symmetric=yes
nat=force_rport is force_rport=yes

and to fix the BYE problem, you will need rewrite_contact=yes

You have not provided any logs for a WebRTC call, nor have you provided any configuration appropriate for WebRTC.

The c= line is where the other party is telling you to send media. Asterisk will start by trying to send media there, but will change to the same place as from where it is receiving media, if rtp_symmetric is set. WebRTC uses much more extensive techniques for getting round NAT; ir will send a large list of possible addresses and the parties will work through them until they find one that works.

Note, in the past, it has been said that you really need a deep understanding of a large number of technologies to successfully use WebRTC, and it sounds like you are expecting it to work out of the box.

17

to fix the BYE problem, you will need rewrite_contact=yes

I’ve added rewrite_contact=yes to the MicroSIP endpoint, it still does not get BYEs.

I never worked with Asterisk before, but now that I need to build a callcenter software I have to learn.
Guides describing how to configure Asterisk WebRTC are short and simple, there are few blocks of configuration and it seems it should work out of the box. And it kind of worked, I could make a call and hear voice within an hour. One way voice..
I know that my case is common and probably there’s a very simple solution. But I’ve spent so much time on this and this is why I’m here to get help from the community.

Here’s everything I have for WebRTC (based from the docs.asterisk.org):
http.conf

[general]
servername=Asterisk
enabled=yes
bindaddr=193.169.240.153
bindport=8088
tlsenable=yes ; enable tls - default no.
tlsbindaddr=193.169.240.153:8089 ; address and port to bind to - default is bindaddr and port 8089.
tlscertfile=/etc/letsencrypt/live/pbx.research.ua/fullchain.pem ; path to the certificate file (.pem) only.
tlsprivatekey=/etc/letsencrypt/live/pbx.research.ua/privkey.pem ; path to private key file (.pem) only.

rtp.conf

[general]
rtpstart=10000
rtpend=65535

pjsip.conf

[wss-transport]
type=transport
protocol=wss
bind=0.0.0.0

[1060]
type=endpoint
aors=1060
auth=1060-auth
transport=wss-transport
context=rtc60
webrtc=yes
disallow=all
allow=ulaw, opus

What else should be there?

I’ve attached the log of the no-sound “hello world” call through WebRTC (which sounds fine on MicroSIP and in the Asterisk recording).
full.txt (37.1 KB)

18

Have you set all three of those I mentioned when I first mentioned rtp_symmetric?

19

yes,

rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes

MicroSIP call to “hello world” response plays but does not hang up.

20

Media is flowing both ways and the BYE has been accepted. by the caller.

This is the confirmation of the BYE;

[Mar 21 16:11:49] VERBOSE[160289] res_pjsip_logger.c: <--- Received SIP response (339 bytes) from WSS:62.216.59.42:64702 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 193.169.240.153:8089;rport;branch=z9hG4bKPj7dc1fe0f-2322-4a65-981d-84276a24e731;alias
To: <sip:1060@mypbx.ua>;tag=6rugs7nt0q
From: <sip:380685978789@mypbx.ua>;tag=dfd85f3e-37cd-4fe2-88ea-31b5e6bab2c0
Call-ID: 991ln4iqr75at0fh9hsu
CSeq: 8121 BYE
Supported: outbound
Content-Length: 0

It looks like it had to fallback to symmetric media, because the client’s ICE candidates didn’t contain anything usable. I don’t know enough about ICE to know if that is a problem, but I’d suggest trying to get a public address in the list:


a=candidate:83796027 1 udp 2122260223 192.168.50.217 63909 typ host generation 0 network-id 1
a=candidate:1058961708 1 udp 2122194687 192.168.56.1 63910 typ host generation 0 network-id 2
a=candidate:2050048675 1 tcp 1518280447 192.168.50.217 9 typ host tcptype active generation 0 network-id 1
a=candidate:1104255924 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 2