One way audio ulaw

Asterisk 18.6.0, two sip devices from different brands, both are using:

  • pjsip endpoints
  • ulaw codec
  • ptime=20
  • no nat
  • pjsip show channelstats: show no packet lost.

One can hear clearly, the other one is completely muted.

I’ve tried disallow=all and allow=one_at_time_each codec.
But seems that only ulaw is managed by both.

If you make a call to voice mail you can hear and talk correctly from both.
Any idea on how can I fix this?

Thank you, BR

is there a firewall between the two phones? sounds like a NAT issue

Confirm that Direct Media is set to No. Capture traffic with tcpdump / Wireshark and report whether audio is being sent to both endpoints.

Thanks for the replies, direct_media is set to no on both extensions, there’s no firewall, only LAN traffic.

Extension are:
1110 (ip.19): start the call > hear 1107
1107 (ip.210): receive the call > cannot hear 1110
PBX: (ip.205): Attached packet recorded with sngrep.

Fix info on the first post: Asterisk PBX 16.16.1

Looking with wireshark > voip it seems that there are wrong timestamps and silences, but cannot find any “how to fix it”.

Please rename the PACKET.tgz to PACKET.pcap
PACKET.tgz (195.2 KB)

Am I missing some informations, to get an opinion about it?

Anyone has opened the tcpdump?

Can you please point me out, where I can address my question to get an help about it?

Where was this trace taken from? We generally prefer a trace taken from Asterisk’s full log file with “pjsip set logger on” enabled.

It surprises me as It is doing things I wouldn’t expect Asterisk to do. Asterisk appears to be reINVITEing with a changed audio port number, but the same IP address, and rejecting a video stream that wasn’t there originally. I’m much more used to analyzing chan_sip logs, but I’m surprised that chan_pjsip would do this. I’m wondering if the trace is being taken the wrong side of some application level gateway, which is messing things up.

If this is really taken from the Asterisk machine, I would want to turn up the debug level on Asterisk, as well as enabling set logger on.

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPj934c1473-fea5-4fe8-ad5b-2e5c09c462f5
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30828 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Type: application/sdp
Content-Length:   259

v=0
o=- 202368400 202368400 IN IP4 10.8.200.205
s=Asterisk
c=IN IP4 10.8.200.205
t=0 0
m=audio 12548 RTP/AVP 0 9 101
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=maxptime:150
a=sendrecv
SIP/2.0 100 Trying
To: <sip:[email protected]>
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30828 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPj934c1473-fea5-4fe8-ad5b-2e5c09c462f5;rport=5060
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

SIP/2.0 180 Ringing
To: <sip:[email protected]>;tag=5031da6099ede0bai0
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30828 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPj934c1473-fea5-4fe8-ad5b-2e5c09c462f5;rport=5060
Contact: casa <sip:[email protected]:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Remote-Party-ID: casa <sip:[email protected]>;screen=yes;party=called
Content-Length: 0

SIP/2.0 200 OK
To: <sip:[email protected]>;tag=5031da6099ede0bai0
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30828 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPj934c1473-fea5-4fe8-ad5b-2e5c09c462f5;rport=5060
Contact: casa <sip:[email protected]:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Remote-Party-ID: casa <sip:[email protected]>;screen=yes;party=called
Content-Length: 257
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 19383579 19383579 IN IP4 10.8.200.210
s=-
c=IN IP4 10.8.200.210
t=0 0
m=audio 19576 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPjf8afe05a-4382-4589-a5cd-3f4357d13c87
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
To: <sip:[email protected]>;tag=5031da6099ede0bai0
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30828 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPje838f5e3-2b72-43f9-945b-c7f580712130
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
To: <sip:[email protected]>;tag=5031da6099ede0bai0
Contact: <sip:[email protected]:5060>
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30829 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Type: application/sdp
Content-Length:   260

v=0
o=- 202368400 202368401 IN IP4 10.8.200.205
s=Asterisk
c=IN IP4 10.8.200.205
t=0 0
m=video 0 RTP/AVP 0 101
m=audio 16442 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
SIP/2.0 200 OK
To: <sip:[email protected]>;tag=5031da6099ede0bai0
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30829 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPje838f5e3-2b72-43f9-945b-c7f580712130;rport=5060
Contact: casa <sip:[email protected]:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 282
Content-Type: application/sdp

v=0
o=- 19384098 19384098 IN IP4 10.8.200.210
s=-
c=IN IP4 10.8.200.210
t=0 0
m=video 0 RTP/AVP 0 101
m=audio 19576 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPj263fc860-1757-40fb-b05c-18c660942406
From: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
To: <sip:[email protected]>;tag=5031da6099ede0bai0
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 30829 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0

BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.210:5060;branch=z9hG4bK-ced80136;rport
From: <sip:[email protected]>;tag=5031da6099ede0bai0
To: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.200.210:5060;rport=5060;received=10.8.200.210;branch=z9hG4bK-ced80136
Call-ID: dcc251ed-8053-43e7-ac66-5e6c14b0dc71
From: <sip:[email protected]>;tag=5031da6099ede0bai0
To: "citofono" <sip:[email protected]>;tag=11761ec8-49e1-40c1-a9fe-e4cc8a5dab73
CSeq: 101 BYE
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0

Thank you for the reply, I’ve recorded the packages with sngrep ( GitHub - irontec/sngrep: Ncurses SIP Messages flow viewer ) this is running on the PBX.
I will turn on the debug and logger and send new log asap.

New data with asterisk log level set to 6 and record from sngrep, all came from the PBX


NEW_PACKET.TGZ (358.1 KB)

asterisk*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (728 bytes) from UDP:10.8.200.19:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.19:5060;rport;branch=z9hG4bK466463663
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 20 INVITE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2111D V4.300.0.0
Expires: 120
Content-Type: application/sdp
Content-Length:   303

v=0
o=0 0 0 IN IP4 10.8.200.19
s=Dahua VT 1.5
c=IN IP4 10.8.200.19
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:1.000000
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 20000 RTP/AVP 101 97 0
a=rtpmap:97 PCM/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (490 bytes) to UDP:10.8.200.19:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.200.19:5060;rport=5060;received=10.8.200.19;branch=z9hG4bK466463663
Call-ID: [email protected]
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]>;tag=z9hG4bK466463663
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1648066590/b05aad37ff86313f484f9c22098054c5",opaque="2bcd91fa5725a15f",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Received SIP request (318 bytes) from UDP:10.8.200.19:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.19:5060;rport;branch=z9hG4bK466463663
Route: <sip:10.8.200.205:5060;lr>
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]>;tag=z9hG4bK466463663
Call-ID: [email protected]
CSeq: 20 ACK
Content-Length: 0


<--- Received SIP request (1001 bytes) from UDP:10.8.200.19:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.19:5060;rport;branch=z9hG4bK249067874
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 21 INVITE
Contact: <sip:1110[email protected]:5060>
Authorization: Digest username="1110", realm="asterisk", nonce="1648066590/b05aad37ff86313f484f9c22098054c5", uri="sip:[email protected]:5060", response="a529666ce96bca87bd9cc8126fbe7802", algorithm=MD5, cnonce="0a4f113b", opaque="2bcd91fa5725a15f", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2111D V4.300.0.0
Expires: 120
Content-Type: application/sdp
Content-Length:   303

v=0
o=0 0 0 IN IP4 10.8.200.19
s=Dahua VT 1.5
c=IN IP4 10.8.200.19
t=0 0
m=video 20001 RTP/AVP 96
a=framerate:1.000000
a=rtpmap:96 H264/90000
a=sendrecv
m=audio 20000 RTP/AVP 101 97 0
a=rtpmap:97 PCM/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (317 bytes) to UDP:10.8.200.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.200.19:5060;rport=5060;received=10.8.200.19;branch=z9hG4bK249067874
Call-ID: [email protected]
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]>
CSeq: 21 INVITE
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


  "User 1110 dialed 1107."
<--- Transmitting SIP request (948 bytes) to UDP:10.8.200.210:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPjb47aaaca-d7f5-420f-8b82-b59c71c0bd2c
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11259 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Type: application/sdp
Content-Length:   261

v=0
o=- 1818060267 1818060267 IN IP4 10.8.200.205
s=Asterisk
c=IN IP4 10.8.200.205
t=0 0
m=audio 16050 RTP/AVP 0 9 101
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=maxptime:150
a=sendrecv

<--- Received SIP response (359 bytes) from UDP:10.8.200.210:5060 --->
SIP/2.0 100 Trying
To: <sip:[email protected]>
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11259 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPjb47aaaca-d7f5-420f-8b82-b59c71c0bd2c;rport=5060
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0


<--- Received SIP response (498 bytes) from UDP:10.8.200.210:5060 --->
SIP/2.0 180 Ringing
To: <sip:[email protected]>;tag=60da2c38f118608bi0
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11259 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPjb47aaaca-d7f5-420f-8b82-b59c71c0bd2c;rport=5060
Contact: casa <sip:[email protected]:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Remote-Party-ID: casa <sip:[email protected]>;screen=yes;party=called
Content-Length: 0


<--- Transmitting SIP response (504 bytes) to UDP:10.8.200.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.8.200.19:5060;rport=5060;received=10.8.200.19;branch=z9hG4bK249067874
Call-ID: [email protected]
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]>;tag=2ea8d831-4655-44ea-a8ed-b3b2774b4c8f
CSeq: 21 INVITE
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Contact: <sip:10.8.200.205:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (879 bytes) from UDP:10.8.200.210:5060 --->
SIP/2.0 200 OK
To: <sip:[email protected]>;tag=60da2c38f118608bi0
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11259 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPjb47aaaca-d7f5-420f-8b82-b59c71c0bd2c;rport=5060
Contact: casa <sip:[email protected]:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Remote-Party-ID: casa <sip:[email protected]>;screen=yes;party=called
Content-Length: 259
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 217984228 217984228 IN IP4 10.8.200.210
s=-
c=IN IP4 10.8.200.210
t=0 0
m=audio 19750 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP request (423 bytes) to UDP:10.8.200.210:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPj82587438-7359-4ba8-96c1-f586dcceb72d
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
To: <sip:[email protected]>;tag=60da2c38f118608bi0
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11259 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Transmitting SIP response (822 bytes) to UDP:10.8.200.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.200.19:5060;rport=5060;received=10.8.200.19;branch=z9hG4bK249067874
Call-ID: [email protected]
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]>;tag=2ea8d831-4655-44ea-a8ed-b3b2774b4c8f
CSeq: 21 INVITE
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.8.200.205:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 0 2 IN IP4 10.8.200.205
s=Asterisk
c=IN IP4 10.8.200.205
t=0 0
m=video 0 RTP/AVP 96
m=audio 18914 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (972 bytes) to UDP:10.8.200.210:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPj39fca9fa-ad68-43cf-b9cc-d72da803f708
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
To: <sip:[email protected]>;tag=60da2c38f118608bi0
Contact: <sip:[email protected]:5060>
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11260 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Type: application/sdp
Content-Length:   262

v=0
o=- 1818060267 1818060268 IN IP4 10.8.200.205
s=Asterisk
c=IN IP4 10.8.200.205
t=0 0
m=video 0 RTP/AVP 0 101
m=audio 14338 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (739 bytes) from UDP:10.8.200.210:5060 --->
SIP/2.0 200 OK
To: <sip:[email protected]>;tag=60da2c38f118608bi0
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11260 INVITE
Via: SIP/2.0/UDP 10.8.200.205:5060;branch=z9hG4bKPj39fca9fa-ad68-43cf-b9cc-d72da803f708;rport=5060
Contact: casa <sip:[email protected]:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 284
Content-Type: application/sdp

v=0
o=- 217984773 217984773 IN IP4 10.8.200.210
s=-
c=IN IP4 10.8.200.210
t=0 0
m=video 0 RTP/AVP 0 101
m=audio 19750 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP request (423 bytes) to UDP:10.8.200.210:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPj72b966ae-3dd4-41b5-a0a5-e11563eecdbf
From: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
To: <sip:[email protected]>;tag=60da2c38f118608bi0
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 11260 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Received SIP request (401 bytes) from UDP:10.8.200.19:5060 --->
ACK sip:10.8.200.205:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.19:5060;rport;branch=z9hG4bK428956071
From: <sip:[email protected]>;tag=1547778300
To: <sip:[email protected]>;tag=2ea8d831-4655-44ea-a8ed-b3b2774b4c8f
Call-ID: [email protected]
CSeq: 21 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2111D V4.300.0.0
Content-Length: 0


<--- Received SIP request (389 bytes) from UDP:10.8.200.210:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.210:5060;branch=z9hG4bK-1010cd87;rport
From: <sip:[email protected]>;tag=60da2c38f118608bi0
To: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0


<--- Transmitting SIP response (373 bytes) to UDP:10.8.200.210:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.200.210:5060;rport=5060;received=10.8.200.210;branch=z9hG4bK-1010cd87
Call-ID: 7717000f-7d9a-430c-9fe9-9889c686e088
From: <sip:[email protected]>;tag=60da2c38f118608bi0
To: "citofono" <sip:[email protected]>;tag=d7401106-631f-413a-87b1-c881939246f6
CSeq: 101 BYE
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Transmitting SIP request (426 bytes) to UDP:10.8.200.19:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.205:5060;rport;branch=z9hG4bKPj33c8782d-d07a-4dc4-8547-00a0947363b5
From: <sip:[email protected]>;tag=2ea8d831-4655-44ea-a8ed-b3b2774b4c8f
To: <sip:[email protected]>;tag=1547778300
Call-ID: [email protected]
CSeq: 5592 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Received SIP response (363 bytes) from UDP:10.8.200.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.200.205:5060;rport=5060;branch=z9hG4bKPj33c8782d-d07a-4dc4-8547-00a0947363b5
From: <sip:[email protected]>;tag=2ea8d831-4655-44ea-a8ed-b3b2774b4c8f
To: <sip:[email protected]>;tag=1547778300
Call-ID: [email protected]
CSeq: 5592 BYE
User-Agent: Dahua UAC/3.0 VTO2111D V4.300.0.0
Content-Length: 0


<--- Received SIP request (387 bytes) from UDP:10.8.200.19:5060 --->
REGISTER sip:10.8.200.205 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.19:5060;rport;branch=z9hG4bK524747287
From: <sip:[email protected]:5060>;tag=2130045037
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2111D V4.300.0.0
Expires: 60
PhoneState: 0
Content-Length: 0


<--- Transmitting SIP response (476 bytes) to UDP:10.8.200.19:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.200.19:5060;rport=5060;received=10.8.200.19;branch=z9hG4bK524747287
Call-ID: [email protected]
From: <sip:[email protected]>;tag=2130045037
To: <sip:[email protected]>;tag=z9hG4bK524747287
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1648066624/6260b71fba87d4486febcd8310cdae46",opaque="5547bbf65a0abd57",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Received SIP request (879 bytes) from UDP:10.8.200.203:48259 --->
REGISTER sip:10.8.200.205 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.203:48259;branch=z9hG4bK.SHwPbJKM8;rport
From: "nubia" <sip:[email protected]>;tag=LyaBoJ3Bh
To: "nubia" <sip:[email protected]>
CSeq: 20 REGISTER
Call-ID: wBruls03FV
Max-Forwards: 70
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: "nubia" <sip:[email protected]:48259;pn-prid=cha9UnbIRlaziJJjV4yCen:APA91bF5r35cvXs7gqdKYNH-GpTb7HetLhzTo_ptO6VkwUxq4v_fiqiuaRCYeqzhSzsMhdycDpqur3o8cGm2D6GNwuQhcxN1_ESS0gj_dOSa3bxljd3BkuGRdD9OK5SN1v6eTEad58PA;pn-provider=fcm;pn-param=929724111839;pn-silent=1;pn-timeout=0;transport=udp>;+sip.instance="<urn:uuid:152e2d70-fedb-00fd-a7c2-d48f91f74135>";+org.linphone.specs="lime"
Expires: 3600
User-Agent: LinphoneAndroid/4.6.3 (Disconnesso) LinphoneSDK/5.1.7-pre.5+18cd243 (release/5.1)


<--- Transmitting SIP response (487 bytes) to UDP:10.8.200.203:48259 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.200.203:48259;rport=48259;received=10.8.200.203;branch=z9hG4bK.SHwPbJKM8
Call-ID: wBruls03FV
From: "nubia" <sip:[email protected]>;tag=LyaBoJ3Bh
To: "nubia" <sip:[email protected]>;tag=z9hG4bK.SHwPbJKM8
CSeq: 20 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1648066625/447e70e7cb66d34d05769ef98b898dd5",opaque="1fb249cc7a5b6f58",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Received SIP request (1160 bytes) from UDP:10.8.200.203:48259 --->
REGISTER sip:10.8.200.205 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.203:48259;branch=z9hG4bK.6menJRzoU;rport
From: "nubia" <sip:[email protected]>;tag=LyaBoJ3Bh
To: "nubia" <sip:[email protected]>
CSeq: 21 REGISTER
Call-ID: wBruls03FV
Max-Forwards: 70
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: "nubia" <sip:[email protected]:48259;pn-prid=cha9UnbIRlaziJJjV4yCen:APA91bF5r35cvXs7gqdKYNH-GpTb7HetLhzTo_ptO6VkwUxq4v_fiqiuaRCYeqzhSzsMhdycDpqur3o8cGm2D6GNwuQhcxN1_ESS0gj_dOSa3bxljd3BkuGRdD9OK5SN1v6eTEad58PA;pn-provider=fcm;pn-param=929724111839;pn-silent=1;pn-timeout=0;transport=udp>;+sip.instance="<urn:uuid:152e2d70-fedb-00fd-a7c2-d48f91f74135>";+org.linphone.specs="lime"
Expires: 3600
User-Agent: LinphoneAndroid/4.6.3 (Disconnesso) LinphoneSDK/5.1.7-pre.5+18cd243 (release/5.1)
Authorization:  Digest realm="asterisk", nonce="1648066625/447e70e7cb66d34d05769ef98b898dd5", algorithm=md5, opaque="1fb249cc7a5b6f58", username="EAFB2F4319C4",  uri="sip:10.8.200.205", response="2589719d6ab53ca30efa44984549fc63", cnonce="T0N9sfzJUIhIc8sg", nc=00000001, qop=auth


<--- Transmitting SIP response (685 bytes) to UDP:10.8.200.203:48259 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.200.203:48259;rport=48259;received=10.8.200.203;branch=z9hG4bK.6menJRzoU
Call-ID: wBruls03FV
From: "nubia" <sip:[email protected]>;tag=LyaBoJ3Bh
To: "nubia" <sip:[email protected]>;tag=z9hG4bK.6menJRzoU
CSeq: 21 REGISTER
Date: Wed, 23 Mar 2022 20:17:05 GMT
Contact: <sip:[email protected]:48259;transport=udp;pn-prid=cha9UnbIRlaziJJjV4yCen:APA91bF5r35cvXs7gqdKYNH-GpTb7HetLhzTo_ptO6VkwUxq4v_fiqiuaRCYeqzhSzsMhdycDpqur3o8cGm2D6GNwuQhcxN1_ESS0gj_dOSa3bxljd3BkuGRdD9OK5SN1v6eTEad58PA;pn-provider=fcm;pn-param=929724111839;pn-silent=1;pn-timeout=0>;expires=3599
Expires: 3600
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0


<--- Received SIP request (651 bytes) from UDP:10.8.200.19:5060 --->
REGISTER sip:10.8.200.205 SIP/2.0
Via: SIP/2.0/UDP 10.8.200.19:5060;rport;branch=z9hG4bK1498111744
From: <sip:[email protected]:5060>;tag=2130045037
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]:5060>
Authorization: Digest username="1110", realm="asterisk", nonce="1648066624/6260b71fba87d4486febcd8310cdae46", uri="sip:10.8.200.205", response="8e145543c445409df9b0f2c02ac08919", algorithm=MD5, cnonce="0a4f113b", opaque="5547bbf65a0abd57", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2111D V4.300.0.0
Expires: 60
PhoneState: 0
Content-Length: 0


<--- Transmitting SIP response (421 bytes) to UDP:10.8.200.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.200.19:5060;rport=5060;received=10.8.200.19;branch=z9hG4bK1498111744
Call-ID: [email protected]
From: <sip:[email protected]200.205>;tag=2130045037
To: <sip:[email protected]>;tag=z9hG4bK1498111744
CSeq: 2 REGISTER
Date: Wed, 23 Mar 2022 20:17:05 GMT
Contact: <sip:[email protected]:5060>;expires=59
Expires: 60
Server: Asterisk PBX 16.16.1~dfsg-1+deb11u1
Content-Length:  0

 

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