I have examine the packet INVITE and have seen the next:
 
AA.AA.AA.AA = public IP address of SER/Mediaproxy Server.
BB.BB.BB.BB = public IP address of endpoint (the endopoint is behind nat)
CC.CC.CC.CC = public IP address of SIP SERVER(carrier)
 
When the SER follows the INVITE message, rewrites the field Contact and
fill it with the public ip address of sip client. Can this to be my problem?
 
In this same message into SDP, in Contact information, the SER change this field BUT write the
ip address two times. Can this a bug?
 
Thank at all,
--
Alberto
 
----------------
INVITE from endpoint to SER:

  Session Initiation Protocol

    Request-Line: INVITE sip:932215863@AA.AA.AA.AA SIP/2.0

        Method: INVITE

        Resent Packet: False

    Message Header

        Via: SIP/2.0/UDP 192.168.100.55:5060;branch=z9hG4bK-63bf38d4;rport

        From: <sip:1000@AA.AA.AA.AA>;tag=c1342f3464087414o0

        To: <sip:932215863@AA.AA.AA.AA>

        Call-ID: d7eca5b4-6a866f94@192.168.100.55

        CSeq: 102 INVITE

        Max-Forwards: 70

        Contact: <sip:1000@192.168.100.55:5060>

        Expires: 240

        User-Agent: Linksys/PAP2-2.0.12(LS)

        Content-Length: 428

        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

        Supported: x-sipura

        Content-Type: application/sdp

    Message body

        Session Description Protocol

            Session Description Protocol Version (v): 0

            Owner/Creator, Session Id (o): - 6735673 6735673 IN IP4 192.168.100.55

                Owner Username: -

                Session ID: 6735673

                Session Version: 6735673

                Owner Network Type: IN

                Owner Address Type: IP4

                Owner Address: 192.168.100.55

            Session Name (s): -

            Connection Information (c): IN IP4 192.168.100.55

                Connection Network Type: IN

                Connection Address Type: IP4

                Connection Address: 192.168.100.55

                   ........................

 

INVITE from SER to SIP SERVER(CARRIER):

Session Initiation Protocol

    Request-Line: INVITE sip:932215863@CC.CC.CC.CC:5060 SIP/2.0

        Method: INVITE

        Resent Packet: False

    Message Header

        Record-Route: <sip:932215863@AA.AA.AA.AA:5060;nat=yes;ftag=c1342f3464087414o0;lr=on>

        Via: SIP/2.0/UDP AA.AA.AA.AA;branch=z9hG4bKa01c.50c2aac6.0

        Via: SIP/2.0/UDP 192.168.100.55:5060;received=BB.BB.BB.BB;branch=z9hG4bK-63bf38d4;rport=60413

        From: <sip:1000@AA.AA.AA.AA>;tag=c1342f3464087414o0

        To: <sip:932215863@AA.AA.AA.AA>

        Call-ID: d7eca5b4-6a866f94@192.168.100.55

        CSeq: 102 INVITE

        Max-Forwards: 16

        Contact: <sip:1000@BB.BB.BB.BB:60413>

        Expires: 240

        User-Agent: Linksys/PAP2-2.0.12(LS)

        Content-Length: 445

        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

        Supported: x-sipura

        Content-Type: application/sdp

    Message body

        Session Description Protocol

            Session Description Protocol Version (v): 0

            Owner/Creator, Session Id (o): - 6735673 6735673 IN IP4 192.168.100.55

                Owner Username: -

                Session ID: 6735673

                Session Version: 6735673

                Owner Network Type: IN

                Owner Address Type: IP4

                Owner Address: 192.168.100.55

            Session Name (s): -

            Connection Information (c): IN IP4 AA.AA.AA.AAAA.AA.AA.AA

                Connection Network Type: IN

                Connection Address Type: IP4

                Connection Address: AA.AA.AA.AAAA.AA.AA.AA     <---------- BUG????????

 

 

----- Original Message -----
From: Alberto
To: serusers@lists.iptel.org
Sent: Thursday, September 22, 2005 10:55 AM
Subject: [Serusers] One path of RTP traffic

Hi,
I have a SER + Mediaproxy. I have not any problem the call between SIP clients (behind or not the NATs)
but when I try to call to PSTN (via cisco) I only have RTP traffic from SIP client to PSTN.
 
Summarizing, the path of rtp traffic would have to be from:

        up:    SIP Client ----> SER ---> GW-PSTN
        down:  SIP Client <---- SER <--- GW-PSTN
 
but, really is:
 
        up:    SIP Client ----> SER ---> GW-PSTN
        down:  SIP Client <------------- GW-PSTN
 
I use the command:
 
     rewritehostport("212.xxx.xxx.xxx:5060");
 
when I match a geographic number.
 
The complete scheme is:
 
    SIP Client ---- NAT --------- SER+Mediaproxy -------- SIP Server --- GWPSTN
 
 
 
Some idea?
Thanks,
 
--
Alberto
 
 
 

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