Asterisk to AWS Voice Connector Unknown PJSIP request Issue

Currently I have an Asterisk Server running in EC2, under a network managed by my company that I’m not able to modify.
I’m connecting the server to a softphone and to a Chime SDK Voice Connector endpoint. The call works fine from end-to-end, there’s audio from both sides from inbound and outbound.
The issue though comes when I’m trying to review the Kinesis logs, there’s basically nothing, there’s a stream happening, yes, but it’s empty. there’s no audio or data that can be recovered (and it’s supposed to work during and after a call). I have a function that records the call and it also works fine.
I’m almost sure that it’s a network or NAT issue (I’m very new to VoIP and PBX servers) but can’t address if it’s from AWS or from my Asterisk configurations.

  • The network is under a subnet that assigns both public and private IPs, the server is under an EIP. The security group, I’ve configured it to allow any kind of traffic from the AWS Chime SDK Voice Connector subnets and any kind of traffic to TCP/443 since it seems to be required for their services: Network configuration and bandwidth requirements - Amazon Chime SDK

Now what I’ve noticed is that in the Voice Connector Termination settings there’s no last OPTIONS ping coming in from Asterisk, even though in the pjsip logs it does send and even return with a 200 OK response from VC.
Ex:

<--- Transmitting SIP request (491 bytes) to UDP:3.80.16.224:5060 --->
OPTIONS sip:ID.voiceconnector.chime.aws SIP/2.0
Via: SIP/2.0/UDP EIP:5060;rport;branch=z9hG4bKPj2bae4957-4aee-4b1e-88e0-ad8126af0a08
From: <sip:+PHONE@EIP>;tag=9824b143-4ae4-4b3e-99b2-24d9c887117b
To: <sip:ID.voiceconnector.chime.aws>
Contact: <sip:+PHONE@EIP:5060>
Call-ID: 486d0cd6-421e-440f-85e6-ccf56effbdac
CSeq: 48315 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.2
Content-Length:  0


<--- Received SIP response (412 bytes) from UDP:3.80.16.224:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP EIP:5060;rport=5060;branch=z9hG4bKPj2bae4957-4aee-4b1e-88e0-ad8126af0a08;received=EIP
From: <sip:+PHONE@EIP>;tag=9824b143-4ae4-4b3e-99b2-24d9c887117b
To: <sip:ID.voiceconnector.chime.aws>;tag=e3273b8d54e5080b56e3d7e8a4aeacaf.27da50c5
Call-ID: 486d0cd6-421e-440f-85e6-ccf56effbdac
CSeq: 48315 OPTIONS
Content-Length: 0

An Invite request:

<--- Transmitting SIP request (1408 bytes) to UDP:3.80.16.244:5060 --->
INVITE sip:+PHONE@ID.voiceconnector.chime.aws SIP/2.0
Via: SIP/2.0/UDP EIP:5060;rport;branch=z9hG4bKPjd8bed7bd-0ef2-4ecc-b99f-b90a2f65ddfc
From: <sip:+PHONE@EIP>;tag=96c4341c-3ca3-45cb-b159-f81162353c6a
To: <sip:+PHONE@ID.voiceconnector.chime.aws>
Contact: <sip:+PHONE@EIP:5060>
Call-ID: 7bdd8f58-6af3-4e9f-b95d-496e2fa04405
CSeq: 472 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 20.5.2
Content-Type: application/sdp
Content-Length:   657

v=0
o=- 1493716567 1493716567 IN IP4 EIP
s=Asterisk
c=IN IP4 EIP
t=0 0
m=audio 28118 RTP/AVP 0 101
a=ice-ufrag:7fa18e132df953882e5e4cda7850f2b0
a=ice-pwd:78788e2468f67f612d7af9cf410001ef
a=candidate:Ha000baa 1 UDP 2130706431 PRIVATE IP 28118 typ host
a=candidate:S321124c7 1 UDP 1694498815 EIP 28118 typ srflx raddr PRIVATE IP rport 28118
a=candidate:Ha000baa 2 UDP 2130706430 PRIVATE IP 28119 typ host
a=candidate:S321124c7 2 UDP 1694498814 EIP 28119 typ srflx raddr PRIVATE IP rport 28119
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

And it also responds with a 200 OK obviously because the call does work!
Don’t really know if the issue could be in the softphone’s configuration (Zoiper) since I don’t know what to move.
Here’s my pjsip.conf:

[udp]
type=transport
protocol=udp
bind=0.0.0.0
external_media_address=EIP
external_signaling_address=EIP
allow_reload=yes

[VoiceConnector]
type=endpoint
context=from-voiceConnector
transport=udp
disallow=all
allow=ulaw
aors=VoiceConnector
direct_media=no
ice_support=yes
force_rport=yes
from_user=PHONE
from_domain=EIP

[VoiceConnector]
type=identify
endpoint=VoiceConnector
match=ID.voiceconnector.chime.aws

[VoiceConnector]
type=aor
contact=sip:ID.voiceconnector.chime.aws
qualify_frequency=30

[PHONE]
type=endpoint
context=from-phone
disallow=all
allow=ulaw
transport=udp
auth=PHONE
aors=PHONE
send_pai=yes
direct_media=no
rewrite_contact=yes
ice_support=yes
force_rport=yes

[PHONE]
type=auth
auth_type=userpass
password=a password
username=PHONE

[PHONE]
type=aor
max_contacts=5

I know this is not a forum for AWS but I’m pretty sure AWS support won’t know how to handle Asterisk logs and configurations, I appreciate and thank you for your support and patience.

Someone who could please help…

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