31 Jul
2008
31 Jul
'08
2:49 p.m.
I am having problems with a NAT device. I have an asterisk server that is hosted on a
public IP and when I connect my Snom phones to it directly audio passes correctly both
ways. When I introduce Kamailio 1.3 into the mix I get one way audio.
I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet.
I would be very grateful for any ideas what the problem could be.
Outgoing calls from Kamailio -> asterisk -> PSTN work fine.
Thank you for your help in advance, it is much appreciated.
=================== config =======================
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# NAT detection
force_rport();
if(nat_uac_test("3"))
{
fix_nated_contact();
}
if (!method=="REGISTER")
{
record_route();
}
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
};
if (uri==myself) {
if (method=="REGISTER")
{
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
};
if (isflagset(5))
{
# set branch flag -- when someone will call this user
# the INVITE will have branch flag 6 set after lookup("location")
setbflag(6);
};
fix_nated_contact();
fix_nated_register();
consume_credentials();
save("location");
exit;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
if(method=="MESSAGE")
{
if (!lookup("location"))
{
sl_send_reply("404", "User Offline");
exit;
}
route(4);
}
if (method=="REGISTER")
{
save("location");
}
route(1);
}
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')){
setbflag(6);
};
if (isflagset(5)||isbflagset(6)) {
route(3);
}
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[3] {
if (is_method("BYE"))
{
unforce_rtp_proxy();
}
else if (is_method("INVITE"))
{
fix_nated_sdp("3");
};
if (isflagset(5))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
}
t_on_reply("1");
}
route[4]
{
if (!t_relay())
{
sl_reply_error();
};
exit;
}
failure_route[2] {
if (isbflagset(6) || isflagset(5))
{
unforce_rtp_proxy();
}
}
onreply_route[1]
{
fix_nated_contact();
if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])")
|| is_method("INVITE"))
{
fix_nated_sdp("2");
force_rport();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
}
==================== SIP PACKETS ===========================
INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP
***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v:
SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name"
<sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m:
<sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102
INVITEUser-Agent: asterisk
Max-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc
appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxx
s=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0
PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101
telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video
17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99
H264/90000a=sendrecv
////////////// PHONE REPLY ////////////////////
SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP
****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP
***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN
*****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID:
*** ID ***
CSeq: 102 INVITEContact: <sip:*** DOMAIN
****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK,
CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk,
hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type:
application/sdpContent-Length: 508
v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***
t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3
gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user
9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34
H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj==
192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20
_________________________________________________________________
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31 Jul
31 Jul
3:32 p.m.
Which direction is the one way audio in - from the Phone to the PSTN or
Vice-Versa - I assume audio outbound is OK and it's the audio returning to
the phone from the PSTN you're missing?
Do you have symmetrical RTP set in the Line settings (under NAT) on the
SNOM?
Is Kamailio inside or outside the NAT?
Neill...;o)
Neill Wilkinson
Principal Consultant
Aeonvista Ltd - opening up new ideas
From: users-bounces(a)lists.kamailio.org
[mailto:users-bounces@lists.kamailio.org] On Behalf Of Ross Beer
Sent: 31 July 2008 12:50
To: users(a)lists.kamailio.org
Subject: [Kamailio-Users] NAT Problems
I am having problems with a NAT device. I have an asterisk server that is
hosted on a public IP and when I connect my Snom phones to it directly audio
passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get
one way audio.
I have fixed the SDP and contacts etc and everything appears to be ok with
the sip packet. I would be very grateful for any ideas what the problem
could be.
Outgoing calls from Kamailio -> asterisk -> PSTN work fine.
Thank you for your help in advance, it is much appreciated.
=================== config =======================
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# NAT detection
force_rport();
if(nat_uac_test("3"))
{
fix_nated_contact();
}
if (!method=="REGISTER")
{
record_route();
}
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
};
if (uri==myself) {
if (method=="REGISTER")
{
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
exit;
};
if (isflagset(5))
{
# set branch flag -- when someone will call this user
# the INVITE will have branch flag 6 set after lookup("location")
setbflag(6);
};
fix_nated_contact();
fix_nated_register();
consume_credentials();
save("location");
exit;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
if(method=="MESSAGE")
{
if (!lookup("location"))
{
sl_send_reply("404", "User Offline");
exit;
}
route(4);
}
if (method=="REGISTER")
{
save("location");
}
route(1);
}
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')){
setbflag(6);
};
if (isflagset(5)||isbflagset(6)) {
route(3);
}
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[3] {
if (is_method("BYE"))
{
unforce_rtp_proxy();
}
else if (is_method("INVITE"))
{
fix_nated_sdp("3");
};
if (isflagset(5))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
}
t_on_reply("1");
}
route[4]
{
if (!t_relay())
{
sl_reply_error();
};
exit;
}
failure_route[2] {
if (isbflagset(6) || isflagset(5))
{
unforce_rtp_proxy();
}
}
onreply_route[1]
{
fix_nated_contact();
if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])")
||
is_method("INVITE"))
{
fix_nated_sdp("2");
force_rport();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
}
==================== SIP PACKETS ===========================
INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0
Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>
Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0
v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060
f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2e
t: <sip:10001*202@** DOMAIN **>
m: <sip:number@**IP**>
i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 69
Date: Thu, 31 Jul 2008 11:38:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
c: application/sdp
l: 422
P-hint: usrloc applied
v=0
o=root 5390 5390 IN IP4 194.xxx.xxx.xxx
s=session
c=IN IP4 194.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 12558 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17918 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
////////////// PHONE REPLY ////////////////////
SIP/2.0 200 Ok
Via: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060
Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>
From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5
To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6n
Call-ID: *** ID ***
CSeq: 102 INVITE
Contact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1
User-Agent: snom370/7.3.7
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, replaces, from-change
Content-Type: application/sdp
Content-Length: 508
v=0
o=root 1569142945 1569142946 IN IP4 192.168.1.20
s=call
c=IN IP4 ** EXTERNAL IP ***
t=0 0
m=audio 10014 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=alt:1 0.9 : user 9kksj== 192.168.1.20 10014
a=sendrecv
m=video 0 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014
a=sendrecv
a=oldmediaip:192.168.1.20
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1 comments
2 participants
participants (2)
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neill.wilkinson＠btinternet.com -
Ross Beer