Multi FS behind same NAT, PRACK goes to wrong port
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 Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
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 Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!
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:
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
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. . ######################################################### Hirdetés (x) RENDELJ MOST! - Háztartási gépeket AKCIÓS áron! Több mint 300 VIDEÓ termékbemutató segít választani, akár 5 ÉV kiterjesztett garanciával rendelhetsz ITT! Sütő-főzőlap szettek, mosógépek, mosogatógépek, tűzhelyek - ORSZÁGOS szállítással a MARKABOLT.hu-tól.
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> wrote:
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
*Hirdetés (x) * Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a kötelező biztosítások kiindulópontja! _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (3)
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Anthony Minessale
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Brian West
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x y