(PJSIP) Outgoing calls have 10-20 second delay before connecting

Asterisk Asterisk SIP
1

I’ve been struggling to figure out the source of this long delay in initiating a call. I’m currently using Telnyx as my Trunk provider. Here is my SIP trace…

I’ve tried several different configurations, but nothing seems to work. I’ve read a little about DNS issues, and I’ve tried a few different DNS servers with no change in the problem. Am I missing something stupid?

**<------------start call------------>**

<--- Received SIP request (1107 bytes) from UDP:74.205.148.150:8508 --->
INVITE sip:5551235002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj9f428c2ba6bd40f0a5b40007d2ec572a
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>
Contact: <sip:6001@74.205.148.150:8508;ob>
Call-ID: 1b88597522634cebbd2c8865acf4522f
CSeq: 25725 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/3.19.8
Content-Type: application/sdp
Content-Length:   479

v=0
o=- 3762596819 3762596819 IN IP4 74.205.148.150
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 39834 RTP/AVP 123 8 0 101
c=IN IP4 74.205.148.150
b=TIAS:64000
a=rtcp:39835 IN IP4 74.205.148.150
a=sendrecv
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=20000;useinbandfec=1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1661800139 cname:79e020d9541e5fb2

<--- Transmitting SIP response (555 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj9f428c2ba6bd40f0a5b40007d2ec572a
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=z9hG4bKPj9f428c2ba6bd40f0a5b40007d2ec572a
CSeq: 25725 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1553629618/a398b9944333aa13ac0427f5db5f875c",opaque="6ad23fbb08cd0b86",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.2.1
Content-Length:  0


<--- Received SIP request (380 bytes) from UDP:74.205.148.150:8508 --->
ACK sip:5551235002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj9f428c2ba6bd40f0a5b40007d2ec572a
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=z9hG4bKPj9f428c2ba6bd40f0a5b40007d2ec572a
Call-ID: 1b88597522634cebbd2c8865acf4522f
CSeq: 25725 ACK
Content-Length:  0


<------------------pause for 10-15 sec.----------------------->


<--- Received SIP request (1404 bytes) from UDP:74.205.148.150:8508 --->
INVITE sip:5551235002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>
Contact: <sip:6001@74.205.148.150:8508;ob>
Call-ID: 1b88597522634cebbd2c8865acf4522f
CSeq: 25726 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/3.19.8
Authorization: Digest username="6001", realm="asterisk", nonce="1553629618/a398b9944333aa13ac0427f5db5f875c", uri="sip:5551235002@52.25.15.38", response="cb1dbaa4a85b226b7c1694aa4de87886", algorithm=md5, cnonce="a264b790e8f842d99d74418d42995c12", opaque="6ad23fbb08cd0b86", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   479

v=0
o=- 3762596819 3762596819 IN IP4 74.205.148.150
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 39834 RTP/AVP 123 8 0 101
c=IN IP4 74.205.148.150
b=TIAS:64000
a=rtcp:39835 IN IP4 74.205.148.150
a=sendrecv
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=20000;useinbandfec=1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1661800139 cname:79e020d9541e5fb2

<--- Transmitting SIP response (357 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>
CSeq: 25726 INVITE
Server: Asterisk PBX 16.2.1
Content-Length:  0


<--- Transmitting SIP request (908 bytes) to UDP:192.76.120.10:5060 --->
INVITE sip:5551235002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPj351448c4-61f1-4787-adab-901b3bc8473d
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>
Contact: <sip:asterisk@172.31.38.120:5060>
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17270 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1100400516 1100400516 IN IP4 172.31.38.120
s=Asterisk
c=IN IP4 172.31.38.120
t=0 0
m=audio 12106 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 (388 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 100 Telnyx trying
Via: SIP/2.0/UDP 172.31.38.120:5060;rport=5060;branch=z9hG4bKPj351448c4-61f1-4787-adab-901b3bc8473d;received=52.25.15.38
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17270 INVITE
Server: kamailio (5.0.7 (x86_64/linux))
Content-Length: 0


<--- Received SIP response (675 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPj351448c4-61f1-4787-adab-901b3bc8473d
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=tcrX4KX7ma9yQ
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17270 INVITE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Proxy-Authenticate: Digest realm="sip.telnyx.com", nonce="da2ca1bc-4c60-4e3e-abcc-8367bab6b4fe", algorithm=MD5, qop="auth"
Content-Length: 0


<--- Transmitting SIP request (409 bytes) to UDP:192.76.120.10:5060 --->
ACK sip:5551235002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPj351448c4-61f1-4787-adab-901b3bc8473d
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=tcrX4KX7ma9yQ
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17270 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Length:  0


<--- Transmitting SIP request (1200 bytes) to UDP:192.76.120.10:5060 --->
INVITE sip:5551235002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>
Contact: <sip:asterisk@172.31.38.120:5060>
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Proxy-Authorization: Digest username="MyUsername", realm="sip.telnyx.com", nonce="da2ca1bc-4c60-4e3e-abcc-8367bab6b4fe", uri="sip:5551235002@sip.telnyx.com:5060", response="40a84db6af088ef033831465c637f118", algorithm=MD5, cnonce="1d3fa52e-b1ae-4785-bced-8fd57989ed3d", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1100400516 1100400516 IN IP4 172.31.38.120
s=Asterisk
c=IN IP4 172.31.38.120
t=0 0
m=audio 12106 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 (388 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 100 Telnyx trying
Via: SIP/2.0/UDP 172.31.38.120:5060;rport=5060;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa;received=52.25.15.38
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 INVITE
Server: kamailio (5.0.7 (x86_64/linux))
Content-Length: 0


<--- Received SIP response (769 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa
Record-Route: <sip:10.255.0.1;transport=tcp;r2=on;lr;ftag=f3478ef0-6abe-4882-a309-366b2ce156c0>
Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=f3478ef0-6abe-4882-a309-366b2ce156c0>
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=ZSpSDUHS6prvH
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 INVITE
Contact: <sip:5551235002@10.15.10.4:5070;transport=tcp>
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0


<--- Transmitting SIP response (545 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=323da61f-3e39-4043-a34c-5c34eb0de536
CSeq: 25726 INVITE
Server: Asterisk PBX 16.2.1
Contact: <sip:172.31.38.120:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (1049 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa
Record-Route: <sip:10.255.0.1;transport=tcp;r2=on;lr;ftag=f3478ef0-6abe-4882-a309-366b2ce156c0>
Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=f3478ef0-6abe-4882-a309-366b2ce156c0>
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=ZSpSDUHS6prvH
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 INVITE
Contact: <sip:5551235002@10.15.10.4:5070;transport=tcp>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 233

v=0
o=FreeSWITCH 1553602452 1553602453 IN IP4 64.16.236.18
s=FreeSWITCH
c=IN IP4 64.16.236.18
t=0 0
m=audio 27190 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=mid:audio

<--- Transmitting SIP response (823 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=323da61f-3e39-4043-a34c-5c34eb0de536
CSeq: 25726 INVITE
Server: Asterisk PBX 16.2.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:172.31.38.120:5060>
Content-Type: application/sdp
Content-Length:   235

v=0
o=- 3762596819 3762596821 IN IP4 52.25.15.38
s=Asterisk
c=IN IP4 52.25.15.38
t=0 0
m=audio 18570 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 response (823 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=323da61f-3e39-4043-a34c-5c34eb0de536
CSeq: 25726 INVITE
Server: Asterisk PBX 16.2.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:172.31.38.120:5060>
Content-Type: application/sdp
Content-Length:   235

v=0
o=- 3762596819 3762596821 IN IP4 52.25.15.38
s=Asterisk
c=IN IP4 52.25.15.38
t=0 0
m=audio 18570 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 request (369 bytes) from UDP:74.205.148.150:8508 --->
CANCEL sip:5551235002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>
Call-ID: 1b88597522634cebbd2c8865acf4522f
CSeq: 25726 CANCEL
User-Agent: MicroSIP/3.19.8
Content-Length:  0


<--- Transmitting SIP response (394 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=323da61f-3e39-4043-a34c-5c34eb0de536
CSeq: 25726 CANCEL
Server: Asterisk PBX 16.2.1
Content-Length:  0


<--- Transmitting SIP response (521 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Call-ID: 1b88597522634cebbd2c8865acf4522f
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=323da61f-3e39-4043-a34c-5c34eb0de536
CSeq: 25726 INVITE
Server: Asterisk PBX 16.2.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Transmitting SIP request (420 bytes) to UDP:192.76.120.10:5060 --->
CANCEL sip:5551235002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Length:  0


<--- Received SIP response (426 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 172.31.38.120:5060;rport=5060;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa;received=52.25.15.38
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=bb4220f0a9320cddd855c6faa18ba481-0928
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 CANCEL
Server: kamailio (5.0.7 (x86_64/linux))
Content-Length: 0


<--- Received SIP request (375 bytes) from UDP:74.205.148.150:8508 --->
ACK sip:5551235002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj4c60e05b3cec4793a73fa032238236db
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=ea4c3a13a519449981bd5d637e271c85
To: <sip:5551235002@52.25.15.38>;tag=323da61f-3e39-4043-a34c-5c34eb0de536
Call-ID: 1b88597522634cebbd2c8865acf4522f
CSeq: 25726 ACK
Content-Length:  0


<--- Received SIP response (515 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=ZSpSDUHS6prvH
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0


<--- Transmitting SIP request (409 bytes) to UDP:192.76.120.10:5060 --->
ACK sip:5551235002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPja8d3de52-b3a7-4647-91d7-64545cc594aa
From: <sip:6001@172.31.38.120>;tag=f3478ef0-6abe-4882-a309-366b2ce156c0
To: <sip:5551235002@sip.telnyx.com>;tag=ZSpSDUHS6prvH
Call-ID: 4def8295-eec9-424d-91d9-74efdb72296f
CSeq: 17271 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Length:  0

And dialplan…

[inbound]
exten => 19706989334,1,Dial(PJSIP/6001)

[users]
exten => _XXX.,1,NoOp(Dialing Out to ${EXTEN})
exten => _XXX.,n,Dial(PJSIP/${EXTEN}@telnyx)
2

The normal reason for such delays is DNS lookups timing out.

For any timing related issue is is almost essentially that you take logs from the log files, where they have time stamps

3

Here you go, here’ a timestamped log.

I used a tcpdump on port 53, but it looks like there aren’t even any dns requests that get sent out until after the log delay?

[2019-03-27 02:15:41] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP request (1106 bytes) from UDP:74.205.148.150:8508 --->
INVITE sip:9709885002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj7dfcb70377b14698a9bdf6fc040923da
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>
Contact: <sip:6001@74.205.148.150:8508;ob>
Call-ID: d7b8481f9db747bfb9a514393c39e275
CSeq: 1106 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/3.19.8
Content-Type: application/sdp
Content-Length:   479

v=0
o=- 3762620142 3762620142 IN IP4 74.205.148.150
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 19610 RTP/AVP 123 8 0 101
c=IN IP4 74.205.148.150
b=TIAS:64000
a=rtcp:19611 IN IP4 74.205.148.150
a=sendrecv
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=20000;useinbandfec=1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1519675804 cname:44e16b37563f0b7c

[2019-03-27 02:15:41] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP response (554 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj7dfcb70377b14698a9bdf6fc040923da
Call-ID: d7b8481f9db747bfb9a514393c39e275
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>;tag=z9hG4bKPj7dfcb70377b14698a9bdf6fc040923da
CSeq: 1106 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1553652941/5b38725649d11a1d595dbdd133fd8571",opaque="404eae5e79a3ea3d",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.2.1
Content-Length:  0


[2019-03-27 02:15:41] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP request (379 bytes) from UDP:74.205.148.150:8508 --->
ACK sip:9709885002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj7dfcb70377b14698a9bdf6fc040923da
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>;tag=z9hG4bKPj7dfcb70377b14698a9bdf6fc040923da
Call-ID: d7b8481f9db747bfb9a514393c39e275
CSeq: 1106 ACK
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP request (1403 bytes) from UDP:74.205.148.150:8508 --->
INVITE sip:9709885002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj9620b95f0da2471eaf6581560593a350
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>
Contact: <sip:6001@74.205.148.150:8508;ob>
Call-ID: d7b8481f9db747bfb9a514393c39e275
CSeq: 1107 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/3.19.8
Authorization: Digest username="6001", realm="asterisk", nonce="1553652941/5b38725649d11a1d595dbdd133fd8571", uri="sip:9709885002@52.25.15.38", response="f5051b8a5058415bdebda652656fec49", algorithm=md5, cnonce="5be42e078b6e41a29445c737897a706f", opaque="404eae5e79a3ea3d", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   479

v=0
o=- 3762620142 3762620142 IN IP4 74.205.148.150
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 19610 RTP/AVP 123 8 0 101
c=IN IP4 74.205.148.150
b=TIAS:64000
a=rtcp:19611 IN IP4 74.205.148.150
a=sendrecv
a=rtpmap:123 opus/48000/2
a=fmtp:123 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=20000;useinbandfec=1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1519675804 cname:44e16b37563f0b7c

[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP response (356 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj9620b95f0da2471eaf6581560593a350
Call-ID: d7b8481f9db747bfb9a514393c39e275
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>
CSeq: 1107 INVITE
Server: Asterisk PBX 16.2.1
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP request (899 bytes) to UDP:192.76.120.10:5060 --->
INVITE sip:9709885002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPjf99015ac-00e3-4e7b-9298-c473c50e4cc9
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>
Contact: <sip:asterisk@172.31.38.120:5060>
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13744 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Type: application/sdp
Content-Length:   223

v=0
o=- 498783839 498783839 IN IP4 (null)
s=Asterisk
c=IN IP4 (null)
t=0 0
m=audio 15308 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

[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP response (395 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 100 Telnyx trying
Via: SIP/2.0/UDP 172.31.38.120:5060;rport=5060;branch=z9hG4bKPjf99015ac-00e3-4e7b-9298-c473c50e4cc9;received=52.25.15.38
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13744 INVITE
Server: kamailio (5.0.7 (x86_64/linux))
Content-Length: 0


[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP request (534 bytes) from UDP:76.25.20.125:44458 --->
REGISTER sip:52.25.15.38:5060;transport=UDP;lr SIP/2.0
Via: SIP/2.0/UDP 76.25.20.125:44458;rport;branch=z9hG4bKPj8d5ddf94-aa1c-476d-8a92-351e5d92c636
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=0e87ecda-0689-4a8f-b5fd-bdc274066f76
To: <sip:6001@52.25.15.38>
Call-ID: f9bf5bc2-6d80-44fa-af29-198cb8b70b3e
CSeq: 55527 REGISTER
User-Agent: Calls
Contact: <sip:6001@76.25.20.125:44458;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP response (565 bytes) to UDP:76.25.20.125:44458 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 76.25.20.125:44458;rport=44458;received=76.25.20.125;branch=z9hG4bKPj8d5ddf94-aa1c-476d-8a92-351e5d92c636
Call-ID: f9bf5bc2-6d80-44fa-af29-198cb8b70b3e
From: <sip:6001@52.25.15.38>;tag=0e87ecda-0689-4a8f-b5fd-bdc274066f76
To: <sip:6001@52.25.15.38>;tag=z9hG4bKPj8d5ddf94-aa1c-476d-8a92-351e5d92c636
CSeq: 55527 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1553652962/cfca579829db7419cad334ed0622785d",opaque="3230fea742067fbb",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.2.1
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP response (682 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPjf99015ac-00e3-4e7b-9298-c473c50e4cc9
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>;tag=p3UQ890N03aNH
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13744 INVITE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Proxy-Authenticate: Digest realm="sip.telnyx.com", nonce="5b8c0fb2-0a9a-4a5c-ae0f-f7a6fed59b5b", algorithm=MD5, qop="auth"
Content-Length: 0


[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP request (416 bytes) to UDP:192.76.120.10:5060 --->
ACK sip:9709885002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPjf99015ac-00e3-4e7b-9298-c473c50e4cc9
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>;tag=p3UQ890N03aNH
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13744 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP request (1191 bytes) to UDP:192.76.120.10:5060 --->
INVITE sip:9709885002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPjfa7b5f4c-7d38-47c3-a8d4-fdfafad52f4a
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>
Contact: <sip:asterisk@172.31.38.120:5060>
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13745 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Proxy-Authorization: Digest username="alex16462", realm="sip.telnyx.com", nonce="5b8c0fb2-0a9a-4a5c-ae0f-f7a6fed59b5b", uri="sip:9709885002@sip.telnyx.com:5060", response="fc8e17256cd2a90ad518f25aa8c0d96e", algorithm=MD5, cnonce="52d67bf1-95c8-41a7-bfc9-3bfe7d18a4c4", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   223

v=0
o=- 498783839 498783839 IN IP4 (null)
s=Asterisk
c=IN IP4 (null)
t=0 0
m=audio 15308 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

[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP response (395 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 100 Telnyx trying
Via: SIP/2.0/UDP 172.31.38.120:5060;rport=5060;branch=z9hG4bKPjfa7b5f4c-7d38-47c3-a8d4-fdfafad52f4a;received=52.25.15.38
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13745 INVITE
Server: kamailio (5.0.7 (x86_64/linux))
Content-Length: 0


[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP request (846 bytes) from UDP:76.25.20.125:44458 --->
REGISTER sip:52.25.15.38:5060;transport=UDP;lr SIP/2.0
Via: SIP/2.0/UDP 76.25.20.125:44458;rport;branch=z9hG4bKPj45c0fb1f-a79c-496b-b78e-b540644ac2f9
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=0e87ecda-0689-4a8f-b5fd-bdc274066f76
To: <sip:6001@52.25.15.38>
Call-ID: f9bf5bc2-6d80-44fa-af29-198cb8b70b3e
CSeq: 55528 REGISTER
User-Agent: Calls
Contact: <sip:6001@76.25.20.125:44458;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="6001", realm="asterisk", nonce="1553652962/cfca579829db7419cad334ed0622785d", uri="sip:52.25.15.38:5060;transport=UDP;lr", response="77f4ad5e1451a98d2bbabc828f6f26cb", algorithm=md5, cnonce="6c682124-49d9-4eea-9c24-de3817b9e01b", opaque="3230fea742067fbb", qop=auth, nc=00000001
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP response (571 bytes) to UDP:76.25.20.125:44458 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 76.25.20.125:44458;rport=44458;received=76.25.20.125;branch=z9hG4bKPj45c0fb1f-a79c-496b-b78e-b540644ac2f9
Call-ID: f9bf5bc2-6d80-44fa-af29-198cb8b70b3e
From: <sip:6001@52.25.15.38>;tag=0e87ecda-0689-4a8f-b5fd-bdc274066f76
To: <sip:6001@52.25.15.38>;tag=z9hG4bKPj45c0fb1f-a79c-496b-b78e-b540644ac2f9
CSeq: 55528 REGISTER
Date: Wed, 27 Mar 2019 02:16:02 GMT
Contact: <sip:6001@74.205.148.150:8508;ob>;expires=119
Contact: <sip:6001@76.25.20.125:44458;ob>;expires=299
Expires: 300
Server: Asterisk PBX 16.2.1
Content-Length:  0


[2019-03-27 02:16:02] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP response (775 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPjfa7b5f4c-7d38-47c3-a8d4-fdfafad52f4a
Record-Route: <sip:10.255.0.1;transport=tcp;r2=on;lr;ftag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc>
Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc>
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>;tag=QcNga5HSXc17c
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13745 INVITE
Contact: <sip:9709885002@10.15.4.4:5070;transport=tcp>
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0


[2019-03-27 02:16:02] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP response (544 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj9620b95f0da2471eaf6581560593a350
Call-ID: d7b8481f9db747bfb9a514393c39e275
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>;tag=9220837a-4ba0-4cf7-87bc-e7fbc6149dc9
CSeq: 1107 INVITE
Server: Asterisk PBX 16.2.1
Contact: <sip:172.31.38.120:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


[2019-03-27 02:16:04] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP response (597 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.31.38.120:5060;received=52.25.15.38;rport=5060;branch=z9hG4bKPjfa7b5f4c-7d38-47c3-a8d4-fdfafad52f4a
Max-Forwards: 69
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>;tag=QcNga5HSXc17c
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13745 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Content-Length: 0


[2019-03-27 02:16:04] VERBOSE[4095] res_pjsip_logger.c: <--- Transmitting SIP request (416 bytes) to UDP:192.76.120.10:5060 --->
ACK sip:9709885002@sip.telnyx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.38.120:5060;rport;branch=z9hG4bKPjfa7b5f4c-7d38-47c3-a8d4-fdfafad52f4a
From: <sip:19706989334@172.31.38.120>;tag=62b615f1-52c1-4cc8-9d84-5ddb8ac2f0cc
To: <sip:9709885002@sip.telnyx.com>;tag=QcNga5HSXc17c
Call-ID: 77c9ecd5-4682-418d-b6fa-c49628e3eadd
CSeq: 13745 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1
Content-Length:  0


[2019-03-27 02:16:04] VERBOSE[4096] res_pjsip_logger.c: <--- Transmitting SIP response (545 bytes) to UDP:74.205.148.150:8508 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 74.205.148.150:8508;rport=8508;received=74.205.148.150;branch=z9hG4bKPj9620b95f0da2471eaf6581560593a350
Call-ID: d7b8481f9db747bfb9a514393c39e275
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>;tag=9220837a-4ba0-4cf7-87bc-e7fbc6149dc9
CSeq: 1107 INVITE
Server: Asterisk PBX 16.2.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=34
Content-Length:  0


[2019-03-27 02:16:04] VERBOSE[1430] res_pjsip_logger.c: <--- Received SIP request (374 bytes) from UDP:74.205.148.150:8508 --->
ACK sip:9709885002@52.25.15.38 SIP/2.0
Via: SIP/2.0/UDP 74.205.148.150:8508;rport;branch=z9hG4bKPj9620b95f0da2471eaf6581560593a350
Max-Forwards: 70
From: <sip:6001@52.25.15.38>;tag=7f367be779df4161a9f12a8363296e4b
To: <sip:9709885002@52.25.15.38>;tag=9220837a-4ba0-4cf7-87bc-e7fbc6149dc9
Call-ID: d7b8481f9db747bfb9a514393c39e275
CSeq: 1107 ACK
Content-Length:  0
4

On the client side delays can also be caused due to delay in ICE gatherings.

5

Would that be likely if I’m not using webrtc? I haven’t even configured anything related to ICE.

6

Can you give some brief about your setup ?

Are you using softphone ? If so, which one ? Can you look at the logs of what is happening on client and asterisk side.