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[Freeswitch-users] Freeswitch hangs up after 30 s when using


 
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woodydickson at gmail.com
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PostPosted: Mon Nov 24, 2008 4:49 am    Post subject: [Freeswitch-users] Freeswitch hangs up after 30 s when using Reply with quote
Hi

I am using Openser as the sip proxy in front of freeswitch. When using Record-Route, Freeswitch hangs of every call after 30 s.

277.32.22.33:5060 is the public ip of openser and 192.168.1.101:5800 is freeswitch's external profile port. Both openser and freeswitch are within the same box.

In the console, I am getting:
2008-11-25 01:31:24 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/1000@61.141.158.178 (1000@61.141.158.178) entering state [terminating]

freeswitch@localhost.localdomain>
freeswitch@localhost.localdomain> 2008-11-25 01:31:24 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/1000@61.141.158.178 (1000@61.141.158.178) entering state [terminated]

Here is the sip trace:

U 192.168.1.101:5800 -> 277.32.22.33:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK022f.a4e32a57.0;received=277.32.22.33.
Via: SIP/2.0/UDP 192.168.1.102:12334;received=121.15.98.134;branch=z9hG4bK-d87543-a5439229f1204a4e-1--d87543-;rport=14392.
Record-Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>.
From: "1000" <sip:1000@277.32.22.33>;tag=c947d86b.
To: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 2 INVITE.
Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk.
Session-Expires: 120;refresher=uas.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 268.
.
v=0.
o=FreeSWITCH 6527595211019529703 806853432324137362 IN IP4 277.32.22.33.
s=FreeSWITCH.
c=IN IP4 277.32.22.33.
t=0 0.
m=audio 11046 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.


U 277.32.22.33:5800 -> 192.168.1.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK022f.a4e32a57.0;received=277.32.22.33.
Via: SIP/2.0/UDP 192.168.1.102:12334;received=121.15.98.134;branch=z9hG4bK-d87543-a5439229f1204a4e-1--d87543-;rport=14392.
Record-Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>.
From: "1000" <sip:1000@277.32.22.33>;tag=c947d86b.
To: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 2 INVITE.
Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk.
Session-Expires: 120;refresher=uas.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 268.
.
v=0.
o=FreeSWITCH 6527595211019529703 806853432324137362 IN IP4 277.32.22.33.
s=FreeSWITCH.
c=IN IP4 277.32.22.33.
t=0 0.
m=audio 11046 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.


U 192.168.1.101:5800 -> 277.32.22.33:5060
BYE sip:1000@121.15.98.134:14392 SIP/2.0.
Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg.
Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>.
Max-Forwards: 70.
From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 107655293 BYE.
Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
.


U 277.32.22.33:5800 -> 192.168.1.101:5060
BYE sip:1000@121.15.98.134:14392 SIP/2.0.
Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg.
Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>.
Max-Forwards: 70.
From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 107655293 BYE.
Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
.


U 192.168.1.101:5800 -> 277.32.22.33:5060
BYE sip:1000@121.15.98.134:14392 SIP/2.0.
Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg.
Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>.
Max-Forwards: 70.
From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 107655293 BYE.
Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
.


U 277.32.22.33:5800 -> 192.168.1.101:5060
BYE sip:1000@121.15.98.134:14392 SIP/2.0.
Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg.
Route: <sip:192.168.1.101;lr=on;ftag=c947d86b>.
Max-Forwards: 70.
From: "0" <sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
To: "1000" <sip:1000@277.32.22.33>;tag=c947d86b.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 107655293 BYE.
Contact: <sip:mod_sofia@277.32.22.33:5800;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
.


U 192.168.1.101:5060 -> 277.32.22.33:5800
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 277.32.22.33:5800;received=277.32.22.33;rport=5800;branch=z9hG4bK10ttgjpr9KeQg.
Record-Route: <sip:192.168.1.101;lr;ftag=4Uve20t8p31Ba>.
Contact: <sip:1000@121.15.98.134:14392>.
To: "1000"<sip:1000@277.32.22.33>;tag=c947d86b.
From: "0"<sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 107655293 BYE.
User-Agent: X-Lite release 1011s stamp 41150.
Content-Length: 0.
P-hint: (3)passed thru onreply_route[1].
.


U 277.32.22.33:5060 -> 192.168.1.101:5800
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 277.32.22.33:5800;received=277.32.22.33;rport=5800;branch=z9hG4bK10ttgjpr9KeQg.
Record-Route: <sip:192.168.1.101;lr;ftag=4Uve20t8p31Ba>.
Contact: <sip:1000@121.15.98.134:14392>.
To: "1000"<sip:1000@277.32.22.33>;tag=c947d86b.
From: "0"<sip:0@277.32.22.33>;tag=4Uve20t8p31Ba.
Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI..
CSeq: 107655293 BYE.
User-Agent: X-Lite release 1011s stamp 41150.
Content-Length: 0.
P-hint: (3)passed thru onreply_route[1].
.


Thanks in advance for any help or suggestion on resolving the problem.

Woody
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ibc at aliax.net
Guest
PostPosted: Mon Nov 24, 2008 4:53 am    Post subject: [Freeswitch-users] Freeswitch hangs up after 30 s when using Reply with quote
El Lunes, 24 de Noviembre de 2008, Woody Dickson escribió:
Quote:
I am using Openser as the sip proxy in front of freeswitch.   When using
Record-Route, Freeswitch hangs of every call after 30 s.

277.32.22.33:5060 is the public ip of openser and 192.168.1.101:5800 is
freeswitch's external profile port.  Both openser and freeswitch are within
the same box. 

This occurs because FreeSwitch is not receiving the ACK after replying 200 OK
(as you can see in your SIP trace).
If a UAS replies 200 Ok it expects to receive ACK during the following 32
seconds, if not, it must understand that there has been a problem and sends a
BYE.

I see an strange network topology in your SIP trace:

FS replies 200 from 192.168.1.101:5800 to 277.32.22.33:5060. What is this
public IP?

Anyway, the problem is that the ACK doesn't arrive to FS, so revise it.

PD: I've OpenSIPS in front of FS and of course FS works well when receiving an
INVITE with Record-Route (of course, because in my case FS also receives the
ACK).

--
Iñaki Baz Castillo

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