I have the following problem which I hope the more knowledgeable on here
could shed some light on:
Between my UA and the SIP server I have set up SRTP/TLS connection on
and I can register normally with that enforced. When somebody (over the
PSTN network using a regular phone number) tries to ring me, I get the
above error.
I did a bit of research and found that possible causes could either be
wrong codec or the enforcement of SRTP (I have it as "Mandatory" in my
UA), but I am still not 100% sure and thought to ask here. The error I
am getting from my UA is shown above, though the underlying problem
seems to be with the "SIP/2.0 488 Not Acceptable Here" message in packet
4 below. This is the full trace of what is happening (I masked my IDs
and the relevant IP subnets):
==========
----------
15:12:06 Packet 1/6, 976 bytes
--->
UDP port 5060 INVITE sip:me at provider.voip
audio->212.x.x.x:12706
UDP port 5060
----------
INVITE sip:me at provider.voip SIP/2.0
Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0
Via: SIP/2.0/UDP
212.x.x.x:5060;received=212.x.x.x;branch=z9hG4bK0679c35a;rport=5060
From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966
To: <sip:me at provider.voip>
Contact: <sip:%2b49xxxx at 212.x.x.x>
Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x
CSeq: 102 INVITE
User-Agent: Provider-VOIP
Max-Forwards: 68
Date: Sat, 06 Oct 2012 15:12:06 CET
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
Thor-Scope: lookup-destination
Thor-Origin: sip:212.x.x.x:5060
v=0
o=root 1888 1888 IN IP4 212.x.x.x
s=session
c=IN IP4 212.x.x.x
t=0 0
m=audio 12706 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
==========
----------
15:12:06 Packet 2/6, 1150 bytes
<---
UDP port 5060 INVITE sip:me at provider.voip:52222
audio->81.x.x.x:57958
UDP port 5060
----------
INVITE sip:me at provider.voip:52222;transport=tls;ob SIP/2.0
Record-Route: <sip:81.x.x.x;lr;ftag=as1012c966;did=05.76f75113>
Via: SIP/2.0/UDP 81.x.x.x;branch=z9hG4bK12fd.65070351.0
Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0
Via: SIP/2.0/UDP
212.x.x.x:5060;received=212.x.x.x;branch=z9hG4bK0679c35a;rport=5060
From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966
To: <sip:me at provider.voip>
Contact: <sip:%2b49xxxx at 212.x.x.x>
Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x
CSeq: 102 INVITE
User-Agent: Provider-VOIP
Max-Forwards: 68
Date: Sat, 06 Oct 2012 15:12:06 CET
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
Thor-Scope: relay-destination
Thor-Target: sip:178.x.x.x:52222;transport=tls
v=0
o=root 1888 1888 IN IP4 212.x.x.x
s=session
c=IN IP4 81.x.x.x
t=0 0
m=audio 57958 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
==========
----------
15:12:06 Packet 3/6, 481 bytes
--->
UDP port 5060 100 Giving a try for INVITE
UDP port 5060
----------
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 81.x.x.x;branch=z9hG4bK12fd.65070351.0
Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0
Via: SIP/2.0/UDP
212.x.x.x:5060;received=212.x.x.x;branch=z9hG4bK0679c35a;rport=5060
From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966
To: <sip:me at provider.voip>
Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x
CSeq: 102 INVITE
Server: SIP Thor on OpenSIPS XS 1.8.0
Content-Length: 0
==========
----------
15:12:06 Packet 4/6, 722 bytes
--->
UDP port 5060 488 Not Acceptable Here for INVITE
UDP port 5060
----------
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 81.x.x.x;branch=z9hG4bK12fd.65070351.0
Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0
Via: SIP/2.0/UDP
212.x.x.x:5060;rport=5060;received=212.x.x.x;branch=z9hG4bK0679c35a
Record-Route:
<sip:85.xx.xx.xx:5061;transport=tls;lr;r2=on;ftag=as1012c966;did=05.c25aa83>
Record-Route: <sip:85.xx.xx.xx;lr;r2=on;ftag=as1012c966;did=05.c25aa83>
Record-Route: <sip:81.x.x.x;lr;ftag=as1012c966;did=05.76f75113>
Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x
From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966
To: <sip:me at provider.voip>;tag=-FBQxyWnTqZHHrOenpJvRCDUnZwCgDYh
CSeq: 102 INVITE
Content-Length: 0
==========
----------
15:12:06 Packet 5/6, 660 bytes
<---
UDP port 5060 488 Not Acceptable Here for INVITE
UDP port 5060
----------
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0
Via: SIP/2.0/UDP
212.x.x.x:5060;rport=5060;received=212.x.x.x;branch=z9hG4bK0679c35a
Record-Route:
<sip:85.xx.xx.xx:5061;transport=tls;lr;r2=on;ftag=as1012c966;did=05.c25aa83>
Record-Route: <sip:85.xx.xx.xx;lr;r2=on;ftag=as1012c966;did=05.c25aa83>
Record-Route: <sip:81.x.x.x;lr;ftag=as1012c966;did=05.76f75113>
Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x
From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966
To: <sip:me at provider.voip>;tag=-FBQxyWnTqZHHrOenpJvRCDUnZwCgDYh
CSeq: 102 INVITE
Content-Length: 0
==========
----------
15:12:06 Packet 6/6, 393 bytes
--->
UDP port 5060 ACK sip:me at provider.voip
UDP port 5060
----------
ACK sip:me at provider.voip SIP/2.0
Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0
From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966
Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x
To: <sip:me at provider.voip>;tag=-FBQxyWnTqZHHrOenpJvRCDUnZwCgDYh
CSeq: 102 ACK
Max-Forwards: 69
User-Agent: SIP Thor on OpenSIPS XS 1.8.0
Content-Length: 0
==========
