VoIP Mailing List Archives

[fs_ask_sy at citromail.hu]

Hy!

There are two different FS behind the same NAT, and there were Reigstration Failures about one or to times a day. The gateway status turned down, then I got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests it.
Now I facing the next problem:
Start the call, all goes right, INVITE goes to port 1352, then after 183 Session progress from port 1352, the PRACK package goes to 5060 instead of 1352, wich messes up the call procedure. Is there anyway to force PRACK to the port to the INVITE has been sent before?

Cheers,
Viktor

[brian at freeswitch.org]

I will bet you that the 183 has no port or the wrong port in the contact.

/b

On Nov 28, 2008, at 8:20 AM, x y wrote:

[fs_ask_sy at citromail.hu]

The whole situation:

xxx.xxx.xxx.xxx:56956---INVITE--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---100-Trying--->xxx.xxx.xxx.xxx:5060
yyy.yyy.yyy.yyy:5060---INVITE--->zzz.zzz.zzz.zzz:1352
zzz.zzz.zzz.zzz:1352---100-Trying--->yyy.yyy.yyy.yyy:5060
zzz.zzz.zzz.zzz:1352---183-Session-Progress--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---PRACK--->zzz.zzz.zzz.zzz:5060
zzz.zzz.zzz.zzz:5060---481-No-Such-Response--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---CANCEL--->zzz.zzz.zzz.zzz:1352
yyy.yyy.yyy.yyy:5060---481-Call/Transaction-Does-Not-Exist--->xxx.xxx.xxx.xxx:5060
xxx.xxx.xxx.xxx:56956---ACK--->yyy.yyy.yyy.yyy:5060
zzz.zzz.zzz.zzz:1352---200-OK--->yyy.yyy.yyy.yyy:5060
zzz.zzz.zzz.zzz:1352---487-Request-Terminated--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---ACK--->zzz.zzz.zzz.zzz:1352

Cheers,
Viktor

[brian at freeswitch.org]

I don't really care where or to the packets come from but the actual contents of the "contact" header in each packet.

/b

On Nov 28, 2008, at 9:50 AM, x y wrote:

[brian at freeswitch.org]

Can you get me a pcap of this scenario? I can almost bet the contact changes in the 183 which causes sofia to send the PRACK to the new port which is the correct behavior. The other option is to turn off 100rel.

/b

On Nov 28, 2008, at 8:50 AM, x y wrote:

[fs_ask_sy at citromail.hu]

Hy!

You were right about the contact in 183, its port 5060 in there.
I've tried turning of 100rel, it seemed to work with calls, but caused some problems with others things, so I would really appreciate if there is another option.
Btw, I have mentioned that, that I had gateway problems too. Setting up ext-ip as stun.freeswitch.org has seemed to work, but after 5 days, the gateway has went down again with the same 503 error. Is there any common in the two issues?

Thx for your advices.

Cheers,
Viktor





################################################################
U xxx.xxx.xxx.xxx:56965 -> yyy.yyy.yyy.yyy:5060
INVITE sip:252252%233619995384@box.net:5060 SIP/2.0.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" <sip:xxx.xxx.xxx.xxx>;tag=7185D258-BB0.
To: <sip:252252%233619995384@box.net>.
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313@xxx.xxx.xxx.xxx.
Supported: timer,100rel.
Min-SE: 1800.
Cisco-Guid: 1234720477-3151434205-2374282417-4237312787.
User: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.
CSeq: 101 INVITE.
Max-Forwards: 10.
Remote-Party-ID: <sip:xxx.xxx.xxx.xxx>;party=calling;screen=yes;privacy=full.
Timestamp: 1227803321.
Contact: <sip:xxx.xxx.xxx.xxx:5060>.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 264.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 5202 8450 IN IP4 xxx.xxx.xxx.xxx.
s=SIP Call.
c=IN IP4 xxx.xxx.xxx.xxx.
t=0 0.
m=audio 16732 RTP/AVP 3 8 101.
c=IN IP4 xxx.xxx.xxx.xxx.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.

#
U yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" <sip:xxx.xxx.xxx.xxx>;tag=7185D258-BB0.
To: <sip:252252%233619995384@box.net>.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313@xxx.xxx.xxx.xxx.
CSeq: 101 INVITE.
Timestamp: 1227803321 0.000388.
User-Agent: agent
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:1352
INVITE sip:3619995384@zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: <sip:usr@yyy.yyy.yyy.yyy:5060;transport=udp>.
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 398.
Remote-Party-ID: "00000000" <sip:00000000@dom>;screen=yes;privacy=full.
.
v=0.
o=FreeSWITCH 6476130113585053783 8141266268953030291 IN IP4 yyy.yyy.yyy.yyy.
s=FreeSWITCH.
c=IN IP4 yyy.yyy.yyy.yyy.
t=0 0.
a=sendrecv.
m=audio 17068 RTP/AVP 3 98 8 9 0 18 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U zzz.zzz.zzz.zzz:1352 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Content-Length: 0.
.

###
U zzz.zzz.zzz.zzz:1352 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: <sip:mod_sofia@zzz.zzz.zzz.zzz:5060;transport=udp>.
RSeq: 2093511444.
User-Agent: agent2
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Require: 100rel.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 264.
.
v=0.
o=FreeSWITCH 6247558966294607749 119302723364474833 IN IP4 zzz.zzz.zzz.zzz.
s=FreeSWITCH.
c=IN IP4 zzz.zzz.zzz.zzz.
t=0 0.
m=audio 24756 RTP/AVP 3 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:5060
PRACK sip:mod_sofia@zzz.zzz.zzz.zzz:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK5DH0ryBH70FSB.
Max-Forwards: 70.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783325 PRACK.
Contact: <sip:usr@yyy.yyy.yyy.yyy:5060;transport=udp>.
RAck: 2093511444 107783324 INVITE.
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Content-Length: 0.
.

#
U zzz.zzz.zzz.zzz:5060 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 481 No such response.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5DH0ryBH70FSB.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783325 PRACK.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:1352
CANCEL sip:3619995384@zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 CANCEL.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
SIP/2.0 481 Call/Transaction Does Not Exist.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" <sip:xxx.xxx.xxx.xxx>;tag=7185D258-BB0.
To: <sip:252252%233619995384@box.net>;tag=cXQ5m2QSmU2QB.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313@xxx.xxx.xxx.xxx.
CSeq: 101 INVITE.
User-Agent: agent
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary.
Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE".
Content-Length: 0.
.

#
U xxx.xxx.xxx.xxx:56965 -> yyy.yyy.yyy.yyy:5060
ACK sip:252252%233619995384@box.net:5060 SIP/2.0.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" <sip:xxx.xxx.xxx.xxx>;tag=7185D258-BB0.
To: <sip:252252%233619995384@box.net>;tag=cXQ5m2QSmU2QB.
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313@xxx.xxx.xxx.xxx.
Max-Forwards: 10.
Content-Length: 0.
CSeq: 101 ACK.
.

#
U zzz.zzz.zzz.zzz:1352 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 CANCEL.
Content-Length: 0.
.

#
U zzz.zzz.zzz.zzz:1352 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:1352
ACK sip:3619995384@zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" <sip:usr@dom;transport=udp>;tag=D6gypX8vH4raQ.
To: <sip:3619995384@zzz.zzz.zzz.zzz:1352>;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 ACK.
Content-Length: 0.
.

#########################################################

[anthony.minessale at g...]

if you enable nat mode on the registrations it will lock the ip:port
make an acl that matches the ip of the client and add the param
apply-nat-acl with the name of the acl you created to your sofia profile
then all calls from that ip will be known to be nat and the port locking code will activate.



On Fri, Nov 28, 2008 at 8:20 AM, x y <fs_ask_sy@citromail.hu (fs_ask_sy@citromail.hu)> wrote: