Hi,
to avoid the bad experience we’ve got with Dahua doorphones, I warn you that both VTO and VTH Dahua are NOT SIP compliant.
i.e. echo test will fail on those devices
Support just said, yes we have a problem, but you’re the only one that notified this, so we will not solve it.
I think this is just wrong and Dahua should not say that product are SIP compliant as they’re not.
BR
In what way are they not compliant? SIP has lots of optional parts.
- from VTH (101) to echo test (43): call ends with DIVERTED
- from VTO (208) to echo test (43): call ends with CALL SETUP
BR
What is Asterisk sending in the Contact header? This looks like an invalid NAT configuration, e…g failing to provide an external address when needed, or failing to list the device network as local, when an external address has been configured.
Thank you for the reply, actually NAT is disabled, but same echo test call from a linphone app is successfull:
Frame 59: 955 bytes on wire (7640 bits), 955 bytes captured (7640 bits)
Linux cooked capture v1
Internet Protocol Version 4, Src: 10.168.5.130, Dst: 10.168.5.138
User Datagram Protocol, Src Port: 5160, Dst Port: 5060
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.168.5.138:5060;branch=z9hG4bK7a1bb354f0071e30571ddb6915c238da;received=10.168.5.138;rport=5060
From: <sip:[email protected]>;tag=30af85278d318eb03fdac1acac937498
SIP from address: sip:[email protected]
SIP from tag: 30af85278d318eb03fdac1acac937498
To: <sip:[email protected]:5160>;tag=as5997db56
SIP to address: sip:[email protected]:5160
SIP to tag: as5997db56
Call-ID: 2021111009591922570556401
[Generated Call-ID: 2021111009591922570556401]
CSeq: 2 INVITE
Server: FPBX-15.0.16.75(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5160>
Contact URI: sip:[email protected]:5160
P-Asserted-Identity: "Test Eco" <sip:*[email protected]>
SIP PAI display info: "Test Eco"
SIP PAI Address: sip:*[email protected]
Content-Type: application/sdp
Content-Length: 312
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 1262148867 1262148867 IN IP4 10.168.5.130
Session Name (s): Asterisk PBX 16.13.0
Connection Information (c): IN IP4 10.168.5.130
Bandwidth Information (b): CT:360
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 23872 RTP/AVP 0 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): maxptime:150
Media Attribute (a): sendrecv
Media Description, name and address (m): video 23526 RTP/AVP 96
Media Attribute (a): rtpmap:96 H264/90000
Media Attribute (a): sendrecv
[Generated Call-ID: 2021111010022743939950434]
Session Initiation Protocol (SIP as raw text)
SIP/2.0 200 OK\r\n
Via: SIP/2.0/UDP 10.168.5.138:5060;branch=z9hG4bK7a1bb354f0071e30571ddb6915c238da;received=10.168.5.138;rport=5060\r\n
From: <sip:[email protected]>;tag=30af85278d318eb03fdac1acac937498\r\n
To: <sip:[email protected]:5160>;tag=as5997db56\r\n
Call-ID: 2021111009591922570556401\r\n
CSeq: 2 INVITE\r\n
Server: FPBX-15.0.16.75(16.13.0)\r\n
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\n
Supported: replaces, timer\r\n
Contact: <sip:[email protected]:5160>\r\n
P-Asserted-Identity: "Test Eco" <sip:*[email protected]>\r\n
Content-Type: application/sdp\r\n
Content-Length: 312\r\n
\r\n
v=0\r\n
o=root 1262148867 1262148867 IN IP4 10.168.5.130\r\n
s=Asterisk PBX 16.13.0\r\n
c=IN IP4 10.168.5.130\r\n
b=CT:360\r\n
t=0 0\r\n
m=audio 23872 RTP/AVP 0 101\r\n
a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=fmtp:101 0-16\r\n
a=maxptime:150\r\n
a=sendrecv\r\n
m=video 23526 RTP/AVP 96\r\n
a=rtpmap:96 H264/90000\r\n
a=sendrecv\r\n
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