Hi,
I have Asterisk on public network and 2 softphones behind NAT. I could only call from Alice to Bob but not from Bob to Alice.
[alice]
type=endpoint
context=testing
disallow=all
allow=ulaw
;allow=t140
;allow=h263p
;allow=h264
;allow=vp8
auth=alice-auth
aors=alice
rtp_symmetric=yes
direct_media=no
force_rport=yes
rewrite_contact=yes
;ice_support=yes
;use_avpf=no
;force_avp=yes
;trust_id_inbound=yes
[alice-auth]
type=auth
auth_type=userpass
username=alice
password=secret123
[alice]
type=aor
max_contacts=10
[bob]
type=endpoint
context=testing
disallow=all
allow=ulaw
;allow=t140
;allow=h263p
;allow=h264
;allow=vp8
auth=bob-auth
aors=bob
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
;ice_support=yes
;use_avpf=no
;force_avp=yes
;trust_id_inbound=yes
[bob-auth]
type=auth
auth_type=userpass
username=bob
password=secret123
[bob]
type=aor
max_contacts=10
Logs:
When calling from Bob to Alice:
-- Executing [6001@testing:1] Dial("PJSIP/bob-00000024", "PJSIP/alice") in new stack
-- Called PJSIP/alice
When calling from Alice to Bob:
[Feb 4 09:13:28] ERROR[31]: pjproject: <?>: sip_transport.c Error processing 740 bytes packet from UDP 5.135.143.170:59858 : PJSIP syntax error exception when parsing 'Request Line' header on line 1 col 12:
INVITE sip: 12074125023@193.40.103.100 SIP/2.0
Via: SIP/2.0/UDP 5.135.143.170:59858;branch=z9hG4bK204701438
Max-Forwards: 70
From: <sip:1186@193.40.103.100>;tag=1190821674
To: <sip: 12074125023@193.40.103.100>
Call-ID: 1040172122-1813647541-1525363005
CSeq: 1 INVITE
Contact: <sip:1186@5.135.143.170:59858>
Content-Type: application/sdp
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: asaxzxccbvcnbxcbzxvssmnnsbscbsds
v=0
o=1186 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11