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

Greger V. Teigre greger at teigre.com
Thu Sep 22 19:58:05 CEST 2005


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)?  
g-(
  ----- Original Message ----- 
  From: Alberto 
  To: Greger V. Teigre ; serusers at lists.iptel.org 
  Sent: Thursday, September 22, 2005 05:08 PM
  Subject: 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 at 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 at 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 at 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 at AA.AA.AA.AA>;tag=c1342f3464087414o0
    >         To: <sip:932215863 at AA.AA.AA.AA>
    >         Call-ID: d7eca5b4-6a866f94 at 192.168.100.55
    >         CSeq: 102 INVITE
    >         Max-Forwards: 70
    >         Contact: <sip:1000 at 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 at CC.CC.CC.CC:5060 SIP/2.0
    >         Method: INVITE
    >         Resent Packet: False
    >     Message Header
    >         Record-Route:
    > <sip:932215863 at 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 at AA.AA.AA.AA>;tag=c1342f3464087414o0
    >         To: <sip:932215863 at AA.AA.AA.AA>
    >         Call-ID: d7eca5b4-6a866f94 at 192.168.100.55
    >         CSeq: 102 INVITE
    >         Max-Forwards: 16
    >         Contact: <sip:1000 at 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 at 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,
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