[Serusers] One path of RTP traffic (possible bug?????)
Alberto
alberto.ipt at telefonica.net
Thu Sep 22 17:08:31 CEST 2005
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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