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<DIV>Hm, a beer is fine ;-) However, maybe I'm the one owing you one. Are you
saying that an unmodified version of the onsip config file has this issue (that
route(4) is called twice)? </DIV>
<DIV>g-(</DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=alberto.ipt@telefonica.net
href="mailto:alberto.ipt@telefonica.net">Alberto</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A title=greger@teigre.com
href="mailto:greger@teigre.com">Greger V. Teigre</A> ; <A
title=serusers@lists.iptel.org
href="mailto:serusers@lists.iptel.org">serusers@lists.iptel.org</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, September 22, 2005 05:08
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Serusers] One path of RTP
traffic (possible bug?????)</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial>Thank you very much....... </FONT><FONT
color=#0000ff><FONT face=Arial color=#000000>I owe you a beer (or
juice)</FONT></FONT></DIV>
<DIV><FONT face=Arial></FONT> </DIV>
<DIV><FONT face=Arial>The problem was the line 272 ( of onSIP SER Getting
Started, PSTN Gateway).Each time we call the function route(5) we have called
previously the route(4) but inside of the function route(5) there are a call
to the function route(4) another time, I called two times the route(4) as your
you said.</FONT></DIV>
<DIV><FONT face=Arial></FONT> </DIV>
<DIV><FONT face=Arial>Best Regards,</FONT></DIV>
<DIV><FONT face=Arial></FONT> </DIV>
<DIV><FONT face=Arial>--</FONT></DIV>
<DIV><FONT face=Arial>Alberto</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>----- Original Message ----- </DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=greger@teigre.com href="mailto:greger@teigre.com">Greger V.
Teigre</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A title=alberto.ipt@telefonica.net
href="mailto:alberto.ipt@telefonica.net">Alberto</A> ; <A
title=serusers@lists.iptel.org
href="mailto:serusers@lists.iptel.org">serusers@lists.iptel.org</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, September 22, 2005 3:40
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Serusers] One path of RTP
traffic (possible bug?????)</DIV>
<DIV><BR></DIV>
<DIV>You probably call fix_nated_sdp() twice in your config.</DIV>
<DIV>g-)</DIV>
<DIV> </DIV>
<DIV>---- Original Message ----<BR>From: Alberto<BR>To: Alberto ; <A
href="mailto:serusers@lists.iptel.org">serusers@lists.iptel.org</A><BR>Sent: Thursday,
September 22, 2005 01:09 PM<BR>Subject: Re: [Serusers] One path of RTP
traffic (possible bug?????)<BR><BR>> I have examine the packet INVITE and
have seen the next:<BR>> <BR>> AA.AA.AA.AA = public IP address of
SER/Mediaproxy Server.<BR>> BB.BB.BB.BB = public IP address of endpoint
(the endopoint is behind<BR>> nat) <BR>> CC.CC.CC.CC = public IP
address of SIP SERVER(carrier)<BR>> <BR>> When the SER follows the
INVITE message, rewrites the field Contact<BR>> and <BR>> fill it with
the public ip address of sip client. Can this to be my<BR>> problem?
<BR>> <BR>> In this same message into SDP, in Contact information, the
SER change<BR>> this field BUT write the <BR>> ip address two times.
Can this a bug?<BR>> <BR>> Thank at all,<BR>> --<BR>>
Alberto<BR>> <BR>> ----------------<BR>> INVITE from endpoint to
SER:<BR>> Session Initiation
Protocol<BR>> Request-Line: INVITE
sip:932215863@AA.AA.AA.AA
SIP/2.0<BR>> Method:
INVITE<BR>> Resent
Packet: False<BR>> Message
Header<BR>> Via:
SIP/2.0/UDP<BR>> 192.168.100.55:5060;branch=z9hG4bK-63bf38d4;rport
<BR>> From:
<sip:1000@AA.AA.AA.AA>;tag=c1342f3464087414o0<BR>>
To:
<sip:932215863@AA.AA.AA.AA><BR>>
Call-ID:
d7eca5b4-6a866f94@192.168.100.55<BR>>
CSeq: 102 INVITE<BR>>
Max-Forwards: 70<BR>>
Contact:
<sip:1000@192.168.100.55:5060><BR>>
Expires: 240<BR>>
User-Agent:
Linksys/PAP2-2.0.12(LS)<BR>>
Content-Length: 428<BR>>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER<BR>> Supported:
x-sipura<BR>>
Content-Type: application/sdp<BR>> Message
body<BR>> Session
Description
Protocol<BR>>
Session Description Protocol Version (v):
0<BR>>
Owner/Creator, Session Id (o): - 6735673 6735673 IN IP4<BR>>
192.168.100.55
<BR>>
Owner Username:
-<BR>>
Session ID:
6735673<BR>>
Session Version:
6735673<BR>>
Owner Network Type:
IN<BR>>
Owner Address Type:
IP4<BR>>
Owner Address:
192.168.100.55<BR>>
Session Name (s):
-<BR>>
Connection Information (c): IN IP4
192.168.100.55<BR>>
Connection Network Type:
IN<BR>>
Connection Address Type:
IP4<BR>>
Connection Address:
192.168.100.55<BR>>
........................<BR>> <BR>> INVITE from SER to SIP
SERVER(CARRIER):<BR>> Session Initiation
Protocol<BR>> Request-Line: INVITE
sip:932215863@CC.CC.CC.CC:5060
SIP/2.0<BR>> Method:
INVITE<BR>> Resent
Packet: False<BR>> Message
Header<BR>>
Record-Route:<BR>>
<sip:932215863@AA.AA.AA.AA:5060;nat=yes;ftag=c1342f3464087414o0;lr=on><BR>>
Via: SIP/2.0/UDP
AA.AA.AA.AA;branch=z9hG4bKa01c.50c2aac6.0<BR>>
Via: SIP/2.0/UDP<BR>>
192.168.100.55:5060;received=BB.BB.BB.BB;branch=z9hG4bK-63bf38d4;rport=60413<BR>>
From:
<sip:1000@AA.AA.AA.AA>;tag=c1342f3464087414o0<BR>>
To:
<sip:932215863@AA.AA.AA.AA><BR>>
Call-ID:
d7eca5b4-6a866f94@192.168.100.55<BR>>
CSeq: 102 INVITE<BR>>
Max-Forwards: 16<BR>>
Contact:
<sip:1000@BB.BB.BB.BB:60413><BR>>
Expires: 240<BR>>
User-Agent:
Linksys/PAP2-2.0.12(LS)<BR>>
Content-Length: 445<BR>>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER<BR>> Supported:
x-sipura<BR>>
Content-Type: application/sdp<BR>> Message
body<BR>> Session
Description
Protocol<BR>>
Session Description Protocol Version (v):
0<BR>>
Owner/Creator, Session Id (o): - 6735673 6735673 IN IP4<BR>>
192.168.100.55
<BR>>
Owner Username:
-<BR>>
Session ID:
6735673<BR>>
Session Version:
6735673<BR>>
Owner Network Type:
IN<BR>>
Owner Address Type:
IP4<BR>>
Owner Address:
192.168.100.55<BR>>
Session Name (s):
-<BR>>
Connection Information (c): IN IP4
AA.AA.AA.AAAA.AA.AA.AA<BR>>
Connection Network Type:
IN<BR>>
Connection Address Type:
IP4<BR>>
Connection Address: AA.AA.AA.AAAA.AA.AA.AA <BR>>
<---------- BUG???????? <BR>> <BR>> <BR>> ----- Original Message
-----<BR>> From: Alberto<BR>> To: serusers@lists.iptel.org<BR>> Sent:
Thursday, September 22, 2005 10:55 AM<BR>> Subject: [Serusers] One path
of RTP traffic<BR>> <BR>> <BR>> Hi,<BR>> I have a SER +
Mediaproxy. I have not any problem the call between<BR>> SIP clients
(behind or not the NATs) <BR>> but when I try to call to PSTN (via cisco)
I only have RTP traffic<BR>> from SIP client to PSTN. <BR>> <BR>>
Summarizing, the path of rtp traffic would have to be from:<BR>>
<BR>>
up: SIP Client ----> SER --->
GW-PSTN<BR>> down:
SIP Client <---- SER <--- GW-PSTN<BR>> <BR>> but, really
is:<BR>> <BR>>
up: SIP Client ----> SER --->
GW-PSTN<BR>> down:
SIP Client <------------- GW-PSTN<BR>> <BR>> I use the
command:<BR>> <BR>>
rewritehostport("212.xxx.xxx.xxx:5060");<BR>> <BR>> when I match a
geographic number.<BR>> <BR>> The complete scheme is:<BR>>
<BR>> SIP Client ---- NAT ---------
SER+Mediaproxy -------- SIP Server<BR>> --- GWPSTN <BR>> <BR>>
<BR>> <BR>> Some idea?<BR>> <BR>>
Thanks,</DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>