SIP/2.0 484 Address Incomplete

Asterisk Asterisk SIP
1

extensions.conf

[default]
exten => 200,1,Answer()
same => n,Playback(demo-congrats)
same => n,Hangup()

http.conf

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
enablestatic=yes      ; without this, you can only send AMI commands, not display 
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.crt
tlsprivatekey=/etc/asterisk/keys/asterisk.key

pjsip.conf

[transport-ws]
type=transport
protocol=ws
bind=0.0.0.0


[webrtc_client]
type=aor
max_contacts=5
remove_existing=yes

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

[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
context=default
disallow=all
allow=opus,ulaw

REGISTER step is successfully done. The INVITE response is coming as 484 Address Incomplete, not finding exact resource for this. Using Asterisk v 18.12.1

Logs

<--- Received SIP request (1548 bytes) from WS:127.0.0.1:42758 --->
INVITE sip:127.0.0.1 SIP/2.0
From: <sip:webrtc_client@127.0.0.1>;tag=f98077d4-29f5-490f-88fe-91716b628e12
CSeq: 8085 INVITE
Call-ID: 222e7771-9689-4545-921b-6a6118402832
Content-Type: application/sdp
Authorization: Digest username="webrtc_client", realm="asterisk", nonce="1656348558/5d09921dbad8abcdc851cb7386bea8f3", uri="sip:200@127.0.0.1", response="9c2c042a8b38992d1675c859050e2ee0", algorithm=MD5, cnonce="q2XR44", qop=auth, nc=00000001
Contact: <sip:webrtc_cleint@127.0.0.1;transport=ws>;expires=200
Via: SIP/2.0/WS 127.0.0.1;branch=z9hG4bK117977e0-d6c3-4ebb-814c-4c4257a8ea32
Supported: replaces, outbound,ice
User-Agent: Pion WebRTC SIP Client
To: <sip:200@127.0.0.1>
Content-Length: 819
Max-Forwards: 70

v=0
o=- 7525629796706269363 1656348558 IN IP4 0.0.0.0
s=-
t=0 0
a=group:BUNDLE 0
a=fingerprint:sha-256 A4:9F:9C:8C:1C:67:D0:F8:9D:F2:95:E8:E6:E2:C2:D1:E3:B6:9F:59:7C:FA:8A:A2:A2:C4:A9:49:B2:58:3A:D1
m=audio 9 UDP/TLS/RTP/SAVPF 111 9 0 8
c=IN IP4 0.0.0.0
a=setup:actpass
a=mid:0
a=ice-ufrag:BbGxrlLGOHGtkttC
a=ice-pwd:yPCSoGeVDxmCuvopuXtJxoDfyBuXCtnx
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ssrc:1186219614 cname:PWLIMWKaoSjdsWyS
a=ssrc:1186219614 msid:PWLIMWKaoSjdsWyS KVqrddcLustIlmBd
a=ssrc:1186219614 mslabel:PWLIMWKaoSjdsWyS
a=ssrc:1186219614 label:KVqrddcLustIlmBd
a=msid:PWLIMWKaoSjdsWyS KVqrddcLustIlmBd
a=sendrecv
a=candidate:79019993 1 udp 1686052607 1.1.1.1 9 typ srflx

<--- Transmitting SIP response (403 bytes) to WS:127.0.0.1:42758 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/WS 127.0.0.1;rport=42758;received=127.0.0.1;branch=z9hG4bK117977e0-d6c3-4ebb-814c-4c4257a8ea32
Call-ID: 222e7771-9689-4545-921b-6a6118402832
From: <sip:webrtc_client@127.0.0.1>;tag=f98077d4-29f5-490f-88fe-91716b628e12
To: <sip:200@127.0.0.1>;tag=56599dde-1e22-431a-bc30-eabcc8743812
CSeq: 8085 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0

What am I not doing correctly here?

2

There is no number dialed in the request URI:

INVITE sip:127.0.0.1 SIP/2.0

It should be:

INVITE sip:200@127.0.0.1 SIP/2.0

What exactly are you doing? Are you building using a SIP implementation at a low level or something?

3

Thank you…This worked, I am getting 100 RINGING response from Asterisk.

Coming to your question. I am building a stack which should basically connect to Asterisk, make SIP calls through webrtc and save the audio to the disk for some testing.

4

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