[Serusers] Problem in SER->SIPH323 communication
Balaji Bapulal Thoguluva
bbthog2 at uky.edu
Wed Jan 21 00:57:30 CET 2004
Hi,
I have this network configuration
SIP Phone (ph.no: 3322) -> SER -> SIPH323 -> router -> Cisco CM ->Cisco Skinny phones(ph.no: 1133).
Their corresponding IP address are
192.168.6.103 -> 192.168.6.100 (port:5060) -> 192.168.6.100 (port 5080 for sip side of SIPH323 converter) -> etc
The problem arises in SER -> SIPH323 communication. SIPH323 doesnot get any SIP recognisable packet that SER sends.
In the SER configuration, I check for uri sip:1133 at 192.168.6.100 and then I forward to port 5080 where SIPH323 listens for SIP side messages.
I have attached the following:
1) My ser configuration 2)ngrep output on lo (loopback interface) 3) ngrep output on eth0.
I would appreciate if you could tell me what is happening and how to solve this problem. I also would like to know if my SER configuartion is correct. I am sure SIPH323 is configured properly.
If you see the ngrep on loopback interface, I could see a message going from 192.168.6.100 -> 192.168.6.100 3:3. This message has some weird characters before the INVITE message. I guess SIPH323 gets this message and doenot respond to it because it is not a valid SIP Invite message. I would like to know why this happens.
ATTACHED FILES:
1) SER.cfg
.
.
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( len_gt(max_len) ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
if(uri=~"^sip:11[0-9]*@192.168.6.100")
{
forward(192.168.6.100, 5080);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
2) ngrep on lo (loopback interface):
interface: lo (127.0.0.0/255.0.0.0)
#
U 192.168.6.100:5060 -> 192.168.6.100:5080
INVITE sip:1133 at 192.168.6.100 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1133 at 192.168.6.100;ftag=925f0b001c06323a74
788-57c59f3f;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch=
0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322 at 192.168.6.100>;tag=925f0b001c06323a74788-57c5
9f3f..To: <sip:1133 at 192.168.6.100>..Call-ID: 000b5f92-63c00
003-69765ab4-289e7e69 at 192.168.6.103..Date: Tue, 20 Jan 2004
23:34:41 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent:
Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip
:3322 at 192.168.6.103:5060..Content-Type: application/sdp..Co
ntent-Length: 225....v=0..o=CiscoSystemsSIP-IPPhone-UserAge
nt 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 19
2.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtp
map:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:
101 0-11..
#
I 192.168.6.100 -> 192.168.6.100 3:3
....E..`.. at .@..d...d...d.....L.7INVITE sip:1133 at 192.168.6.1
00 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1133 at 192.1
68.6.100;ftag=925f0b001c06323a74788-57c59f3f;lr=on>..Via: S
IP/2.0/UDP 192.168.6.100;branch=0..Via: SIP/2.0/UDP 192.168
.6.103:5060..From: "User ID Balaji" <sip:3322 at 192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133 at 192.168
.6.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69 at 192.1
68.6.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 IN
VITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accep
t: application/sdp..C
#
U 192.168.6.100:5060 -> 192.168.6.100:5080
INVITE sip:1133 at 192.168.6.100 SIP/2.0..Max-Forwards: 10..Re
cord-Route: <sip:1133 at 192.168.6.100;ftag=925f0b001c06323a74
788-57c59f3f;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch=
0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Bala
ji" <sip:3322 at 192.168.6.100>;tag=925f0b001c06323a74788-57c5
9f3f..To: <sip:1133 at 192.168.6.100>..Call-ID: 000b5f92-63c00
003-69765ab4-289e7e69 at 192.168.6.103..Date: Tue, 20 Jan 2004
23:34:41 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent:
Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip
:3322 at 192.168.6.103:5060..Content-Type: application/sdp..Co
ntent-Length: 225....v=0..o=CiscoSystemsSIP-IPPhone-UserAge
nt 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 19
2.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtp
map:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:
101 0-11..
...It repeats the same messages
3) ngrep on eth0:
interface: eth0 (192.168.6.0/255.255.255.0)
###############
U 192.168.6.103:49834 -> 192.168.6.100:5060
INVITE sip:1133 at 192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1
68.6.103:5060..From: "User ID Balaji" <sip:3322 at 192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133 at 192.168.6
.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69 at 192.168.6
.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..
Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli
cation/sdp..Contact: sip:3322 at 192.168.6.103:5060..Content-Typ
e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem
sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP
Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0
8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-11..
#
U 192.168.6.103:49834 -> 192.168.6.100:5060
INVITE sip:1133 at 192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1
68.6.103:5060..From: "User ID Balaji" <sip:3322 at 192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133 at 192.168.6
.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69 at 192.168.6
.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..
Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli
cation/sdp..Contact: sip:3322 at 192.168.6.103:5060..Content-Typ
e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem
sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP
Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0
8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-11..
#
U 192.168.6.103:49834 -> 192.168.6.100:5060
INVITE sip:1133 at 192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1
68.6.103:5060..From: "User ID Balaji" <sip:3322 at 192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133 at 192.168.6
.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69 at 192.168.6
.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..
Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli
cation/sdp..Contact: sip:3322 at 192.168.6.103:5060..Content-Typ
e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem
sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP
Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0
8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-11..
###
...it repeats these messages
Thanks,
Balaji
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