Issue with Calls Between Indoor and Outdoor Video Door Phones Using PJSIP

Asterisk Asterisk SIP
1

Hi,
I am using Asterisk with the following pjsip.conf configuration to connect two mobile phones (using Linphone app) and two video door phones (indoor and outdoor):

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:7457
external_media_address=ip
external_signaling_address=ip

[400]
type = aor
max_contacts = 1
remove_existing=yes

[400]
type = auth
auth_type = userpass
username = 400
password = 400

[400]
type = endpoint
transport=transport-udp
aors = 400
auth = 400
context = 1
disallow = all
allow=ulaw,alaw,h264
force_rport = yes
rtp_symmetric = yes
direct_media=no
ice_support = no

[500]
type = aor
max_contacts = 1
remove_existing=yes

[500]
type = auth
auth_type = userpass
username = 500
password = 500

[500]
type = endpoint
transport=transport-udp
aors = 500
auth = 500
context = 1
disallow = all
allow=ulaw,alaw,h264
force_rport = yes
rtp_symmetric = yes
direct_media= no
ice_support = no

Issue:

  • Mobile devices (700 and 900) can call each other successfully.
  • However, when attempting to call from the outdoor video door phone (500) to the indoor video door phone (400), the call does not go through.

What I’ve Tried:

  • Verified the registration status of all devices.
  • Confirmed that context=1 is applied to all endpoints.
  • Ensured allow=ulaw,alaw,h264 for video and audio compatibility.

Questions:

  1. Could there be an issue with the codec negotiation (e.g., h264) or NAT settings?
  2. Is there something specific to video door phones I should configure differently in pjsip.conf?
  3. What additional logs or debugging steps should I take to diagnose this issue?

Any help or insights would be greatly appreciated!
Thank you!

2

You haven’t specified the local networks. If the outdoor phone is not on the same subnet as the Asterisk box interface, it will be considered to have a public address.

Also, “the call does not go through” is not enough detail; you need to provide verbose full logs, possible with pjsip set logger on in effect.

3

Thank you for your input. My Asterisk server is hosted on an AWS EC2 instance, and as such, it is not part of a traditional local network setup. Instead, it communicates over public IP.

Could you please clarify what changes I need to make in this scenario?

After updating the settings as mentioned below, I am able to receive the audio call from the outdoor unit to the indoor unit. However, the video is not displaying. Could you kindly assist me with resolving this issue?

[global]
externip = public-ip-of-ec2-instance
localnet = 172.31.0.0/16

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:7457
external_media_address=public-ip-of-ec2-instance
external_signaling_address=public-ip-of-ec2-instance
;localnet = 172.31.0.0/16

[transport-ws]
type=transport
protocol=ws
bind=0.0.0.0:8088
external_media_address=public-ip-of-ec2-instance
external_signaling_address=public-ip-of-ec2-instance

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
external_media_address=public-ip-of-ec2-instance
external_signaling_address=public-ip-of-ec2-instance


[400]
type=endpoint
context=1
disallow=all
allow=ulaw,alaw
allow=h264
auth=400auth
aors=400
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
max_audio_streams=10
max_video_streams=10

[400]
type=aor
max_contacts=1
remove_existing=yes
remove_unavailable=yes


[500]
type=endpoint
context=1
disallow=all
allow=ulaw,alaw
allow=h264
auth=500auth
aors=500
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
max_audio_streams=10
max_video_streams=10


[500auth]
type=auth
auth_type=userpass
password=500
username=500

[500]
type=aor
max_contacts=1
remove_existing=yes
remove_unavailable=yes```
4

You need to provide detailed protocol logging, e.g. “pjsip set logger on” in effect. However, my guess is that both phones are inside NAT and only one is originating video, making it impossible for symmetric RTP to discover the public address of the other one. If that is the case, I suspect the only solution is bring Asterisk and the video devices into the same private network by using a VPN as part of that network.

5

Hi,

I do not see where you told the endpoints what transport to use. Also where is [400auth]?

6

This is the log i am receiving.

<--- Transmitting SIP response (478 bytes) to UDP:122.171.21.77:11124 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=11124;received=122.171.21.77;branch=z9hG4bK973059607
Call-ID: 1419131707@192.168.1.102
From: <sip:400@192.168.1.102>;tag=492906047
To: <sip:500@mydomain.com>;tag=1255c213-b6cb-49c6-91ef-7d32d7913934
CSeq: 2 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=18
Content-Length:  0


<--- Received SIP request (726 bytes) from UDP:122.171.21.77:4949 --->
INVITE sip:500@mydomain.com:7457 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK1822617208
From: <sip:400@192.168.1.102>;tag=1575816285
To: <sip:500@mydomain.com:7457>
Call-ID: 1660007894@192.168.1.102
CSeq: 3 INVITE
User-Agent: YATE/5.5.0
Contact: <sip:400@192.168.1.102:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 290

v=0
o=yate 1732054283 1732054283 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP response (468 bytes) to UDP:122.171.21.77:4949 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=4949;received=122.171.21.77;branch=z9hG4bK1822617208
Call-ID: 1660007894@192.168.1.102
From: <sip:400@192.168.1.102>;tag=1575816285
To: <sip:500@mydomain.com>;tag=z9hG4bK1822617208
CSeq: 3 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1733285335/ec5f57e2cc0dd574eed7e6c42a8df733",opaque="4f89bd3f3c66c3fa",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (362 bytes) from UDP:122.171.21.77:4949 --->
ACK sip:500@mydomain.com:7457 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK1822617208
From: <sip:400@192.168.1.102>;tag=1575816285
To: <sip:500@mydomain.com:7457>;tag=z9hG4bK1822617208
Call-ID: 1660007894@192.168.1.102
CSeq: 3 ACK
Max-Forwards: 20
Contact: <sip:400@192.168.1.102:5060>
User-Agent: YATE/5.5.0
Content-Length: 0


<--- Received SIP request (1023 bytes) from UDP:122.171.21.77:4949 --->
INVITE sip:500@mydomain.com:7457 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK2072967811
From: <sip:400@192.168.1.102>;tag=1575816285
To: <sip:500@mydomain.com:7457>
Call-ID: 1660007894@192.168.1.102
User-Agent: YATE/5.5.0
Contact: <sip:400@192.168.1.102:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
CSeq: 4 INVITE
Authorization: Digest username="400", realm="asterisk", nonce="1733285335/ec5f57e2cc0dd574eed7e6c42a8df733", uri="sip:500@mydomain.com:7457", response="0a72acad906434106de19ceb2cacfb73", algorithm=MD5, opaque="4f89bd3f3c66c3fa", qop=auth, nc=00000002, cnonce="d17d39a211f3d2f253c4c79d54e5a2a2"
Content-Type: application/sdp
Content-Length: 290

v=0
o=yate 1732054283 1732054283 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP response (294 bytes) to UDP:122.171.21.77:4949 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=4949;received=122.171.21.77;branch=z9hG4bK2072967811
Call-ID: 1660007894@192.168.1.102
From: <sip:400@192.168.1.102>;tag=1575816285
To: <sip:500@mydomain.com>
CSeq: 4 INVITE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- Executing [500@1:1] Set("PJSIP/400-00000006", "SIP_CODEC=h264") in new stack
    -- Executing [500@1:2] Dial("PJSIP/400-00000006", "PJSIP/500") in new stack
    -- Called PJSIP/500
<--- Transmitting SIP request (1090 bytes) to UDP:122.171.21.77:30040 --->
INVITE sip:500@122.171.21.77:30040;line=e33f640faecc4ba SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7457;rport;branch=z9hG4bKPjd89e2e80-e8d1-479c-a758-09bcb8e8cd07
From: <sip:400@ip-1>;tag=98b1ab77-c640-43c5-a81c-9599b9fb4699
To: <sip:500@122.171.21.77;line=e33f640faecc4ba>
Contact: <sip:asterisk@public-ip-ec2:7457>
Call-ID: 79df5679-a659-4e70-a9b5-89d8521b5b3c
CSeq: 18340 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 20.5.0
Content-Type: application/sdp
Content-Length:   383

v=0
o=- 1720868121 1720868121 IN IP4 public-ip-ec2
s=Asterisk
c=IN IP4 public-ip-ec2
t=0 0
m=audio 16238 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 18482 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

<--- Received SIP response (356 bytes) from UDP:122.171.21.77:30040 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP public-ip-ec2:7457;rport=7457;branch=z9hG4bKPjd89e2e80-e8d1-479c-a758-09bcb8e8cd07
From: <sip:400@ip-1>;tag=98b1ab77-c640-43c5-a81c-9599b9fb4699
To: <sip:500@122.171.21.77;line=e33f640faecc4ba>
Call-ID: 79df5679-a659-4e70-a9b5-89d8521b5b3c
CSeq: 18340 INVITE
User-Agent: HKVS/2.0.0
Content-Length: 0


<--- Received SIP response (422 bytes) from UDP:122.171.21.77:30040 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP public-ip-ec2:7457;rport=7457;branch=z9hG4bKPjd89e2e80-e8d1-479c-a758-09bcb8e8cd07
From: <sip:400@ip-1>;tag=98b1ab77-c640-43c5-a81c-9599b9fb4699
To: <sip:500@122.171.21.77;line=e33f640faecc4ba>;tag=700005272
Call-ID: 79df5679-a659-4e70-a9b5-89d8521b5b3c
CSeq: 18340 INVITE
Contact: <sip:500@192.168.1.7:5060>
User-Agent: HKVS/2.0.0
Content-Length: 0


<--- Received SIP response (408 bytes) from UDP:122.171.21.77:30040 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP public-ip-ec2:7457;rport=7457;branch=z9hG4bKPjd89e2e80-e8d1-479c-a758-09bcb8e8cd07
From: <sip:400@ip-1>;tag=98b1ab77-c640-43c5-a81c-9599b9fb4699
To: <sip:500@122.171.21.77;line=e33f640faecc4ba>;tag=700005272
Call-ID: 79df5679-a659-4e70-a9b5-89d8521b5b3c
CSeq: 18340 INVITE
Contact: <sip:500@192.168.1.7:5060>
User-Agent: HKVS/2.0.0
Content-Length: 0


    -- PJSIP/500-00000007 is ringing
<--- Transmitting SIP response (483 bytes) to UDP:122.171.21.77:4949 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=4949;received=122.171.21.77;branch=z9hG4bK2072967811
Call-ID: 1660007894@192.168.1.102
From: <sip:400@192.168.1.102>;tag=1575816285
To: <sip:500@mydomain.com>;tag=889a3b75-b3bd-4f25-a830-42c17a30827b
CSeq: 4 INVITE
Server: Asterisk PBX 20.5.0
Contact: <sip:public-ip-ec2:7457>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP request (494 bytes) from UDP:122.171.21.77:30040 --->
MESSAGE sip:asterisk@public-ip-ec2:7457 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;rport;branch=z9hG4bK2054362730
From: <sip:500@122.171.21.77;line=e33f640faecc4ba>;tag=700005272
To: <sip:400@ip-1>;tag=98b1ab77-c640-43c5-a81c-9599b9fb4699
Call-ID: 79df5679-a659-4e70-a9b5-89d8521b5b3c
CSeq: 2 MESSAGE
Contact: <sip:500@192.168.1.7:5060>
Content-Type: text/plain
Max-Forwards: 70
User-Agent: HKVS/2.0.0
Content-Length:    48

<locknumXML>
<lockNum>0</lockNum>
</locknumXML>

<--- Transmitting SIP response (368 bytes) to UDP:122.171.21.77:30040 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.7:5060;rport=30040;received=122.171.21.77;branch=z9hG4bK2054362730
Call-ID: 79df5679-a659-4e70-a9b5-89d8521b5b3c
From: <sip:500@122.171.21.77;line=e33f640faecc4ba>;tag=700005272
To: <sip:400@ip-1>;tag=98b1ab77-c640-43c5-a81c-9599b9fb4699
CSeq: 2 MESSAGE
Server: Asterisk PBX 20.5.0
Content-Length:  0
7

Here is my [400auth]

[400auth]
type=auth
auth_type=userpass
password=400
username=400

Thank you for pointing that out—I had forgotten to include transport-udp earlier but have since added it. Unfortunately, the issue persists, and calls from the outdoor phone (500) to the indoor phone (400) still fail.

8

The log is incomplete (the outgoing leg is still ringing and hasn’t been answered, and also starts with the end of a previous failed call.

The A side is definitely behind NAT, and is not compensating for it, so it will be impossible for Asterisk to send media to it, until it has sent media to Asterisk.

At the point where the logs cut off, the B side hasn’t told Asterisk how it expects to receive media, so it is impossible for Asterisk to send media to it, even if it send media to Asterisk.

9

I have recently configured early media to send video on incoming calls to my indoor unit. Here’s the configuration I added:

exten => _XXX,1,Progress()          ; Send early media indication
same => n,Dial(PJSIP/${EXTEN})      ; Dial the extension
same => n,Hangup()

The issue is that I receive the video properly on the first call, but the video gets stuck after that. The video will not appear in subsequent calls, even after waiting for some time. When I make another call, the video works again for the first call but not for subsequent ones.

And my log is here.

<--- Received SIP request (731 bytes) from UDP:122.171.21.77:5490 --->
INVITE sip:500@domain.com:7456 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK376425127
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com:7456>
Call-ID: 1155894894@192.168.1.102
CSeq: 66 INVITE
User-Agent: YATE/5.5.0
Contact: <sip:400@192.168.1.102:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 290

v=0
o=yate 1732083215 1732083215 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP response (472 bytes) to UDP:122.171.21.77:5490 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5490;received=122.171.21.77;branch=z9hG4bK376425127
Call-ID: 1155894894@192.168.1.102
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com>;tag=z9hG4bK376425127
CSeq: 66 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1733314264/fcc67b59b1244c3508e9acf7f942d901",opaque="52303f9877b4f756",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (366 bytes) from UDP:122.171.21.77:5490 --->
ACK sip:500@domain.com:7456 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK376425127
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com:7456>;tag=z9hG4bK376425127
Call-ID: 1155894894@192.168.1.102
CSeq: 66 ACK
Max-Forwards: 20
Contact: <sip:400@192.168.1.102:5060>
User-Agent: YATE/5.5.0
Content-Length: 0


<--- Received SIP request (1028 bytes) from UDP:122.171.21.77:5490 --->
INVITE sip:500@domain.com:7456 SIP/2.0
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK913601322
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com:7456>
Call-ID: 1155894894@192.168.1.102
User-Agent: YATE/5.5.0
Contact: <sip:400@192.168.1.102:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
CSeq: 67 INVITE
Authorization: Digest username="400", realm="asterisk", nonce="1733314264/fcc67b59b1244c3508e9acf7f942d901", uri="sip:500@domain.com:7456", response="60aa5efe05dea8a01056ac32eb9f5782", algorithm=MD5, opaque="52303f9877b4f756", qop=auth, nc=00000026, cnonce="bb6df7770b57954461aa5d6b92c2cbef"
Content-Type: application/sdp
Content-Length: 290

v=0
o=yate 1732083215 1732083215 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP response (299 bytes) to UDP:122.171.21.77:5490 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5490;received=122.171.21.77;branch=z9hG4bK913601322
Call-ID: 1155894894@192.168.1.102
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com>
CSeq: 67 INVITE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- Executing [500@1:1] NoOp("PJSIP/400-0000000f", "Starting call flow for extension 500") in new stack
    -- Executing [500@1:2] Progress("PJSIP/400-0000000f", "") in new stack
    -- Executing [500@1:3] Set("PJSIP/400-0000000f", "CALLERID(num)=1234567890") in new stack
    -- Executing [500@1:4] Set("PJSIP/400-0000000f", "MUSICCLASS=default") in new stack
    -- Executing [500@1:5] Dial("PJSIP/400-0000000f", "PJSIP/500,30") in new stack
<--- Transmitting SIP response (890 bytes) to UDP:122.171.21.77:5490 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5490;received=122.171.21.77;branch=z9hG4bK913601322
Call-ID: 1155894894@192.168.1.102
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
CSeq: 67 INVITE
Server: Asterisk PBX 20.5.0
Contact: <sip:public-ip-ec2:7456>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   359

v=0
o=- 1732083215 1732083217 IN IP4 public-ip-ec2
s=Asterisk
c=IN IP4 public-ip-ec2
t=0 0
m=audio 18262 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
m=video 19734 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

    -- Called PJSIP/500
<--- Transmitting SIP request (1100 bytes) to UDP:122.171.21.77:11917 --->
INVITE sip:500@122.171.21.77:11917;line=6884d5276018d03 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj23012c1a-b33a-4720-ae73-382543fdfc1d
From: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
To: <sip:500@122.171.21.77;line=6884d5276018d03>
Contact: <sip:asterisk@public-ip-ec2:7456>
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 9004 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 20.5.0
Content-Type: application/sdp
Content-Length:   381

v=0
o=- 617644431 617644431 IN IP4 public-ip-ec2
s=Asterisk
c=IN IP4 public-ip-ec2
t=0 0
m=audio 10780 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 18180 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

<--- Received SIP response (368 bytes) from UDP:122.171.21.77:11917 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj23012c1a-b33a-4720-ae73-382543fdfc1d
From: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
To: <sip:500@122.171.21.77;line=6884d5276018d03>
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 9004 INVITE
User-Agent: HKVS/2.0.0
Content-Length: 0


<--- Received SIP response (434 bytes) from UDP:122.171.21.77:11917 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj23012c1a-b33a-4720-ae73-382543fdfc1d
From: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
To: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 9004 INVITE
Contact: <sip:500@192.168.1.7:5060>
User-Agent: HKVS/2.0.0
Content-Length: 0


<--- Received SIP request (507 bytes) from UDP:122.171.21.77:11917 --->
MESSAGE sip:asterisk@public-ip-ec2:7456 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;rport;branch=z9hG4bK1424391282
From: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
To: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 2 MESSAGE
Contact: <sip:500@192.168.1.7:5060>
Content-Type: text/plain
Max-Forwards: 70
User-Agent: HKVS/2.0.0
Content-Length:    48

<locknumXML>
<lockNum>0</lockNum>
</locknumXML>

<--- Received SIP response (420 bytes) from UDP:122.171.21.77:11917 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj23012c1a-b33a-4720-ae73-382543fdfc1d
From: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
To: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 9004 INVITE
Contact: <sip:500@192.168.1.7:5060>
User-Agent: HKVS/2.0.0
Content-Length: 0


<--- Transmitting SIP response (381 bytes) to UDP:122.171.21.77:11917 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.7:5060;rport=11917;received=122.171.21.77;branch=z9hG4bK1424391282
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
From: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
To: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
CSeq: 2 MESSAGE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- PJSIP/500-00000010 is ringing
<--- Transmitting SIP response (890 bytes) to UDP:122.171.21.77:5490 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5490;received=122.171.21.77;branch=z9hG4bK913601322
Call-ID: 1155894894@192.168.1.102
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
CSeq: 67 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:public-ip-ec2:7456>
Content-Type: application/sdp
Content-Length:   359

v=0
o=- 1732083215 1732083217 IN IP4 public-ip-ec2
s=Asterisk
c=IN IP4 public-ip-ec2
t=0 0
m=audio 18262 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
m=video 19734 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:24818 --->
OPTIONS sip:700@122.171.21.77:24818 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjc4d47fdb-b8c8-4202-afc0-72075b1cddb5
From: <sip:700@private-ip-ec2>;tag=33288bc0-3326-469e-8198-2ec717df2a86
To: <sip:700@122.171.21.77>
Contact: <sip:700@public-ip-ec2:7456>
Call-ID: 34c08bab-fd6a-41d5-bc59-f8f952c2aab1
CSeq: 48528 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP response (738 bytes) from UDP:122.171.21.77:11917 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj23012c1a-b33a-4720-ae73-382543fdfc1d
From: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
To: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 9004 INVITE
Contact: <sip:500@192.168.1.7:5060>
Content-Type: application/sdp
User-Agent: HKVS/2.0.0
Content-Length:   288

v=0
o=Q32022856 0 0 IN IP4 192.168.1.7
s=Talk session
c=IN IP4 192.168.1.7
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9654 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=4D001F
a=recvonly

<--- Transmitting SIP request (423 bytes) to UDP:122.171.21.77:11917 --->
ACK sip:500@122.171.21.77:11917 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj7f024e66-1a5c-4fef-a0fd-bf0c5ec7a542
From: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
To: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 9004 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


    -- PJSIP/500-00000010 answered PJSIP/400-0000000f
<--- Transmitting SIP response (924 bytes) to UDP:122.171.21.77:5490 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5490;received=122.171.21.77;branch=z9hG4bK913601322
Call-ID: 1155894894@192.168.1.102
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
CSeq: 67 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:public-ip-ec2:7456>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   359

v=0
o=- 1732083215 1732083217 IN IP4 public-ip-ec2
s=Asterisk
c=IN IP4 public-ip-ec2
t=0 0
m=audio 18262 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
m=video 19734 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=4D001F
a=sendrecv

    -- Channel PJSIP/500-00000010 joined 'simple_bridge' basic-bridge <8dad196d-68de-4ed3-9f14-f2850abd2b1c>
    -- Channel PJSIP/400-0000000f joined 'simple_bridge' basic-bridge <8dad196d-68de-4ed3-9f14-f2850abd2b1c>
<--- Received SIP request (676 bytes) from UDP:122.171.21.77:5490 --->
ACK sip:public-ip-ec2:7456 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK1348605793
From: "400" <sip:400@192.168.1.102>;tag=703436105
To: <sip:public-ip-ec2:7456>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
Call-ID: 1155894894@192.168.1.102
CSeq: 67 ACK
Max-Forwards: 20
Contact: <sip:400@192.168.1.102:5060>
Authorization: Digest username="400", realm="asterisk", nonce="1733314264/fcc67b59b1244c3508e9acf7f942d901", uri="sip:500@domain.com:7456", response="60aa5efe05dea8a01056ac32eb9f5782", algorithm=MD5, opaque="52303f9877b4f756", qop=auth, nc=00000026, cnonce="bb6df7770b57954461aa5d6b92c2cbef"
User-Agent: YATE/5.5.0
Content-Length: 0


<--- Transmitting SIP request (1023 bytes) to UDP:122.171.21.77:5490 --->
INVITE sip:400@122.171.21.77:5490 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj443b878b-12c9-43e5-812e-9e7c47201f76
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Contact: <sip:public-ip-ec2:7456>
Call-ID: 1155894894@192.168.1.102
CSeq: 12379 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 20.5.0
Content-Type: application/sdp
Content-Length:   359

v=0
o=- 1732083215 1732083218 IN IP4 public-ip-ec2
s=Asterisk
c=IN IP4 public-ip-ec2
t=0 0
m=audio 18262 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
m=video 19734 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=4D001F
a=recvonly

<--- Received SIP response (364 bytes) from UDP:122.171.21.77:5490 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj443b878b-12c9-43e5-812e-9e7c47201f76;received=public-ip-ec2
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12379 INVITE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (779 bytes) from UDP:122.171.21.77:5490 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj443b878b-12c9-43e5-812e-9e7c47201f76;received=public-ip-ec2
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12379 INVITE
Server: YATE/5.5.0
Contact: <sip:400@192.168.1.102:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 290

v=0
o=yate 1732083215 1732083215 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 9654 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4D001F; packetization-mode=1

<--- Transmitting SIP request (384 bytes) to UDP:122.171.21.77:5490 --->
ACK sip:400@122.171.21.77:5490 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj13e5bb21-678a-4b85-b81d-26ff68e81108
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12379 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (506 bytes) from UDP:122.171.21.77:11917 --->
MESSAGE sip:asterisk@public-ip-ec2:7456 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;rport;branch=z9hG4bK777480472
From: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
To: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 3 MESSAGE
Contact: <sip:500@192.168.1.7:5060>
Content-Type: text/plain
Max-Forwards: 70
User-Agent: HKVS/2.0.0
Content-Length:    48

<locknumXML>
<lockNum>0</lockNum>
</locknumXML>

<--- Transmitting SIP response (380 bytes) to UDP:122.171.21.77:11917 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.7:5060;rport=11917;received=122.171.21.77;branch=z9hG4bK777480472
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
From: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
To: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
CSeq: 3 MESSAGE
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (469 bytes) to UDP:122.171.21.77:5490 --->
MESSAGE sip:400@122.171.21.77:5490 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj531f8674-b576-45ce-859e-701e5f1a5682
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12380 MESSAGE
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: text/plain
Content-Length:    48

<locknumXML>
<lockNum>0</lockNum>
</locknumXML>

<--- Received SIP response (365 bytes) from UDP:122.171.21.77:5490 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj531f8674-b576-45ce-859e-701e5f1a5682;received=public-ip-ec2
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12380 MESSAGE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (461 bytes) from UDP:122.171.21.77:5490 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj531f8674-b576-45ce-859e-701e5f1a5682;received=public-ip-ec2
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12380 MESSAGE
Server: YATE/5.5.0
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: text/plain
Content-Length: 1


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:24818 --->
OPTIONS sip:700@122.171.21.77:24818 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjc4d47fdb-b8c8-4202-afc0-72075b1cddb5
From: <sip:700@private-ip-ec2>;tag=33288bc0-3326-469e-8198-2ec717df2a86
To: <sip:700@122.171.21.77>
Contact: <sip:700@public-ip-ec2:7456>
Call-ID: 34c08bab-fd6a-41d5-bc59-f8f952c2aab1
CSeq: 48528 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (423 bytes) to UDP:122.171.21.77:20154 --->
OPTIONS sip:400@122.171.21.77:20154 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjefc2a31a-97da-4af7-a915-69bf8ed1149e
From: <sip:400@private-ip-ec2>;tag=511e0cef-ef4d-4bbf-aa45-ad4358c7f862
To: <sip:400@122.171.21.77>
Contact: <sip:400@public-ip-ec2:7456>
Call-ID: 3e0faa06-8dde-4b9f-8cd8-69bd078823c5
CSeq: 2715 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (423 bytes) to UDP:122.171.21.77:20154 --->
OPTIONS sip:400@122.171.21.77:20154 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjefc2a31a-97da-4af7-a915-69bf8ed1149e
From: <sip:400@private-ip-ec2>;tag=511e0cef-ef4d-4bbf-aa45-ad4358c7f862
To: <sip:400@122.171.21.77>
Contact: <sip:400@public-ip-ec2:7456>
Call-ID: 3e0faa06-8dde-4b9f-8cd8-69bd078823c5
CSeq: 2715 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


    -- Removed contact 'sip:400@122.171.21.77:20154;x-ast-orig-host=192.168.1.102:5060' from AOR '400' due to expiration
    -- Removed contact 'sip:700@122.171.21.77:24818' from AOR '700' due to expiration
    -- Removed contact 'sip:900@122.171.21.77:11912' from AOR '900' due to expiration
<--- Transmitting SIP request (423 bytes) to UDP:122.171.21.77:20154 --->
OPTIONS sip:400@122.171.21.77:20154 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjefc2a31a-97da-4af7-a915-69bf8ed1149e
From: <sip:400@private-ip-ec2>;tag=511e0cef-ef4d-4bbf-aa45-ad4358c7f862
To: <sip:400@122.171.21.77>
Contact: <sip:400@public-ip-ec2:7456>
Call-ID: 3e0faa06-8dde-4b9f-8cd8-69bd078823c5
CSeq: 2715 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:24818 --->
OPTIONS sip:700@122.171.21.77:24818 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjc4d47fdb-b8c8-4202-afc0-72075b1cddb5
From: <sip:700@private-ip-ec2>;tag=33288bc0-3326-469e-8198-2ec717df2a86
To: <sip:700@122.171.21.77>
Contact: <sip:700@public-ip-ec2:7456>
Call-ID: 34c08bab-fd6a-41d5-bc59-f8f952c2aab1
CSeq: 48528 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (423 bytes) to UDP:122.171.21.77:20154 --->
OPTIONS sip:400@122.171.21.77:20154 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjefc2a31a-97da-4af7-a915-69bf8ed1149e
From: <sip:400@private-ip-ec2>;tag=511e0cef-ef4d-4bbf-aa45-ad4358c7f862
To: <sip:400@122.171.21.77>
Contact: <sip:400@public-ip-ec2:7456>
Call-ID: 3e0faa06-8dde-4b9f-8cd8-69bd078823c5
CSeq: 2715 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:11912 --->
OPTIONS sip:900@122.171.21.77:11912 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj3d19c6b6-1b87-4b8a-b3e3-9c49503176ac
From: <sip:900@private-ip-ec2>;tag=f74eed9d-f028-4129-98a0-5514fdfaf7da
To: <sip:900@122.171.21.77>
Contact: <sip:900@public-ip-ec2:7456>
Call-ID: 2c65b3f2-38e8-45f7-996f-f66524f1cb63
CSeq: 62662 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:11912 --->
OPTIONS sip:900@122.171.21.77:11912 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj3d19c6b6-1b87-4b8a-b3e3-9c49503176ac
From: <sip:900@private-ip-ec2>;tag=f74eed9d-f028-4129-98a0-5514fdfaf7da
To: <sip:900@122.171.21.77>
Contact: <sip:900@public-ip-ec2:7456>
Call-ID: 2c65b3f2-38e8-45f7-996f-f66524f1cb63
CSeq: 62662 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:11912 --->
OPTIONS sip:900@122.171.21.77:11912 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj3d19c6b6-1b87-4b8a-b3e3-9c49503176ac
From: <sip:900@private-ip-ec2>;tag=f74eed9d-f028-4129-98a0-5514fdfaf7da
To: <sip:900@122.171.21.77>
Contact: <sip:900@public-ip-ec2:7456>
Call-ID: 2c65b3f2-38e8-45f7-996f-f66524f1cb63
CSeq: 62662 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:24818 --->
OPTIONS sip:700@122.171.21.77:24818 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjc4d47fdb-b8c8-4202-afc0-72075b1cddb5
From: <sip:700@private-ip-ec2>;tag=33288bc0-3326-469e-8198-2ec717df2a86
To: <sip:700@122.171.21.77>
Contact: <sip:700@public-ip-ec2:7456>
Call-ID: 34c08bab-fd6a-41d5-bc59-f8f952c2aab1
CSeq: 48528 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (420 bytes) from UDP:122.171.21.77:11917 --->
BYE sip:asterisk@public-ip-ec2:7456 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;rport;branch=z9hG4bK522722188
From: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
To: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
CSeq: 4 BYE
Contact: <sip:500@192.168.1.7:5060>
Max-Forwards: 70
User-Agent: HKVS/2.0.0
Content-Length: 0


<--- Transmitting SIP response (370 bytes) to UDP:122.171.21.77:11917 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:5060;rport=11917;received=122.171.21.77;branch=z9hG4bK522722188
Call-ID: aae48e57-c948-4a64-b8d8-d5756cb31e96
From: <sip:500@122.171.21.77;line=6884d5276018d03>;tag=213771234
To: "400" <sip:1234567890@private-ip-ec2>;tag=8fd1690c-a995-4b5d-b11d-e1895b0952a2
CSeq: 4 BYE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- Channel PJSIP/500-00000010 left 'simple_bridge' basic-bridge <8dad196d-68de-4ed3-9f14-f2850abd2b1c>
    -- Channel PJSIP/400-0000000f left 'simple_bridge' basic-bridge <8dad196d-68de-4ed3-9f14-f2850abd2b1c>
  == Spawn extension (1, 500, 5) exited non-zero on 'PJSIP/400-0000000f'
<--- Transmitting SIP request (408 bytes) to UDP:122.171.21.77:5490 --->
BYE sip:400@122.171.21.77:5490 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj15333259-2e88-4b54-95fe-c78cbe1b50da
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12381 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP response (361 bytes) from UDP:122.171.21.77:5490 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj15333259-2e88-4b54-95fe-c78cbe1b50da;received=public-ip-ec2
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12381 BYE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (459 bytes) from UDP:122.171.21.77:5490 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP public-ip-ec2:7456;rport=7456;branch=z9hG4bKPj15333259-2e88-4b54-95fe-c78cbe1b50da;received=public-ip-ec2
From: <sip:500@domain.com>;tag=f632ae2a-1cdc-4477-be33-130295b5da40
To: "400" <sip:400@192.168.1.102>;tag=703436105
Call-ID: 1155894894@192.168.1.102
CSeq: 12381 BYE
P-RTP-Stat: PS=0,OS=0,PR=521,OR=122400,PL=0
Server: YATE/5.5.0
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Length: 0


<--- Transmitting SIP request (423 bytes) to UDP:122.171.21.77:20154 --->
OPTIONS sip:400@122.171.21.77:20154 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPjefc2a31a-97da-4af7-a915-69bf8ed1149e
From: <sip:400@private-ip-ec2>;tag=511e0cef-ef4d-4bbf-aa45-ad4358c7f862
To: <sip:400@122.171.21.77>
Contact: <sip:400@public-ip-ec2:7456>
Call-ID: 3e0faa06-8dde-4b9f-8cd8-69bd078823c5
CSeq: 2715 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Transmitting SIP request (424 bytes) to UDP:122.171.21.77:11912 --->
OPTIONS sip:900@122.171.21.77:11912 SIP/2.0
Via: SIP/2.0/UDP public-ip-ec2:7456;rport;branch=z9hG4bKPj3d19c6b6-1b87-4b8a-b3e3-9c49503176ac
From: <sip:900@private-ip-ec2>;tag=f74eed9d-f028-4129-98a0-5514fdfaf7da
To: <sip:900@122.171.21.77>
Contact: <sip:900@public-ip-ec2:7456>
Call-ID: 2c65b3f2-38e8-45f7-996f-f66524f1cb63
CSeq: 62662 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0
10

On Wednesday 04 December 2024 at 13:28:41, Vignesh-AK via Asterisk Community
wrote:

The issue is that I receive the video properly on the first call, but the
video gets stuck after that. The video will not appear in subsequent
calls, even after waiting for some time. When I make another call, the
video works again for the first call but not for subsequent ones.

I cannot understand “the video will not appear in subsequent calls” and also
“when I make another call, the video works again”.

What’s the difference between “subsequent calls” and “making another call”?

Antony.

–
I bought a book on memory techniques, but I’ve forgotten where I put it.

                                               Please reply to the list;
                                                     please *don't* CC me.
11

By ‘making another call,’ I meant making a call after several hours. The issue occurs when I make the initial call and the video works, but after that, It won’t work. It only works again after making a call several hours later.

12

Hi,
I attempted to install Asterisk locally using the same configuration, including the identical pjsip.conf, extensions.conf, and other settings. On my local setup, I was able to successfully receive video on the indoor device.

here is my rtp.conf

[general]
rtpstart=10000
rtpend=20000
rtpchecksums=no
dtmftimeout=3000
rtcpinterval=5000
rtptimeout=60
rtpholdtimeout=300
strictrtp=no
icesupport=yes
stunaddr=stun.l.google.com:19302
[ice_host_candidates]
13

If they are all on the same LAN, the addresses in the SDP will be correct, and won’t have to be learned from the actual traffic. As such, that is not inconsistent with a symmetric media stand off.

14

I’m a beginner in working with SIP servers and Asterisk, and I need some guidance on how to enable video communication through my cloud SIP server.

I understand that video support in SIP can be complex, especially when dealing with NAT traversal and media routing. Here are the specifics of my setup:

  • I’m using Asterisk as the SIP server in AWS EC2 Instance.
  • My SIP clients (softphones or devices) need to communicate via video.
  • The clients might be behind restrictive NAT or in different networks, and I have the possibility of deploying a TURN server (like coturn) if necessary.
  • I’m running Asterisk in the cloud and need to ensure that video is properly routed between clients.

I would appreciate it if you could provide guidance on the following:

  1. What configuration changes do I need to make in Asterisk to enable video (e.g., enabling video codecs, configuring SDP, etc.)?
  2. Do I need any specific settings for media traversal, such as STUN, TURN, or other settings to ensure video communication works across different networks or NATs?
  3. How do I ensure proper media (video) routing when clients are in different locations or behind NAT?

Thanks in advance for your help! I look forward to your suggestions.

15

Is this an access control project? I have worked on several similar projects

16

No, It is a home automation project like Home Assistant.

17

@Vignesh-AK Do not create another thread for this same issue.

18

Thank you for pointing that out. I appreciate the guidance and will ensure to keep all updates and related queries in the same thread moving forward. My apologies if creating another thread caused any inconvenience.

19

Okay, is this soultion allows residents of multi-family, multiple dwelling unit buildings the ability to see visitors before allowing access into the building?

20

I’d assume it was a hobby (or similar) project for a single family house.