29 Aug
2014
29 Aug
'14
8:42 p.m.
Please consider the following SIP packet exchange, as seen by a tcpdump running on
201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170:
IP 198.58.101.75.5060 > 201.234.196.170.5060
INVITE sip:*43@201.234.196.170:5060 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
Max-Forwards: 70
From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9
To: <sip:*43@201.234.196.170:5060>
Contact: <sip:9002@198.58.101.75:5060>
Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Fri, 29 Aug 2014 18:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 521741684 521741684 IN IP4 198.58.101.75
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 198.58.101.75
t=0 0
m=audio 16426 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
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9
To: <sip:*43@201.234.196.170:5060>
Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060
CSeq: 102 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
IP 201.234.196.170.5060 > 198.58.101.75.5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
Record-Route:
<sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9
To: <sip:*43@201.234.196.170:5060>;tag=as2798a3b9
Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*43@127.0.0.1:5080;alias=127.0.0.1~5080~1>
Content-Type: application/sdp
Require: timer
Content-Length: 305
v=0
o=root 159029581 159029581 IN IP4 201.234.196.170
s=Asterisk PBX 11.12.0
c=IN IP4 201.234.196.170
t=0 0
m=audio 18446 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
a=nortpproxy:yes
According to a strict interpretation of the SIP RFC, which address should the machine at
198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract
said address? I am trying to understand an issue of a missing ACK between
201.234.196.17x and a different public IP, with the only difference that the other IP is
not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I
want to know whether the packets correctly indicate the address for the ACK,
or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.