6 Apr
2005
6 Apr
'05
6:58 a.m.
I changed the line modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock") to modparam("nathelper",
"rtpproxy_sock", "udp:localhost:22222") and started the rtpproxy as
./rtpproxy -s udp from the relevant directory and this resulted in a series of
"rtpp_command: no response from rtpproxy" and rtpproxy temporarily
disabled" errors. If I return to the original modparam and start it as ./rtpproxy
then it works but like I said when the private client rings the public client, I get
"ERROR: send_rtpp_command: cant read reply from a RTP Proxy".
Any further ideas? Has anyone on the mailing list experienced this? I am using the script
given in the onsip getting started doc for 0.9.0. but am using ser 0.8.14.
BR,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
See inline.
Thank you for that Greger. I have altered my script so
that it
exactly mimics the one in the onsip document besides the has_totag
and fix_nated register. All is good when I ring from a private phone
to a public phone i.e. the audio is very clear and the following
messages are in /var/log.
ERROR: extract_body: message body has length zero
ERROR: force_rtp_proxy2: cant extract body from the message.
I assume this is because of the 200 OK to a register message where
theres no sdp?? Is this correct?
That's correct. You will find code in the example configs where we test for an empty
body before calling force_rtp_proxy.
However when I try to phone from public into private I
get:
ERROR: send_rtpp_command: cant read reply from a RTP Proxy.
I find this confusing because I know the rtpproxy is working.
This means that rtpproxy is not responding to a particular message. I have heard some
people have had problems with the socket based communication. I only use UDP. This is what
you do to set up udp (22222 is default port):
modparam("nathelper", "rtpproxy_sock",
"udp:localhost:22222")
rtpproxy must be started with -s udp:*
g-)
BR
Vivienne.
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Yes, you can use fix_nated_contact instead. It is not entirely
RFC-compliant, but that's what you have in 0.8.14.
The has_totag() only tests to see if the INVITE has a To header,
which means that it is in-dialog and thus is a re-INVITE. An INVITE
will normally not have loose routing unless you have another SIP
proxy forwarding an INVITE to you (in which case you should assume
that the other proxy handles NAT and thus not trigger NAT-related
code). You can safely remove the has_totag() if you use
force_rtp_proxy("l")
g-)
---- Original Message ----
From: Vivienne Curran
To: Greger V. Teigre ; serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 02:25 PM
Subject: Re: [Serusers] Contact Header and SDP not rewritten
> Greger,
>
> Since fix_nated_register does not exist with 0.8.14, will
> fix_nated_contact do instead? Also if I am leaving out the
> has_totag() at the start of the script, will this greatly effect its
> functionality?
>
> Thank you,
> Vivienne
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