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(a)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(a)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(a)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(a)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(a)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(a)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,