Re: [Serusers] One path of RTP traffic (possible bug?????)

Thank you very much....... I owe you a beer (or juice)

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.

Best Regards,

-- Alberto

----- Original Message ----- From: Greger V. Teigre To: Alberto ; serusers@lists.iptel.org Sent: Thursday, September 22, 2005 3:40 PM Subject: Re: [Serusers] One path of RTP traffic (possible bug?????)

You probably call fix_nated_sdp() twice in your config. g-)

---- Original Message ---- From: Alberto To: Alberto ; serusers@lists.iptel.org Sent: Thursday, September 22, 2005 01:09 PM Subject: Re: [Serusers] One path of RTP traffic (possible bug?????)

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,