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<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 ;
serusers@lists.iptel.org<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></BODY></HTML>