No we don't have such a feature because the media and signaling are separated by design of SIP.<br>It would be a horrible assumption to always think the media was coming from the same IP as the SIP messages.<br>And even then you still need the port #<br>
<br><br><div class="gmail_quote">On Tue, Mar 23, 2010 at 1:00 PM, Fabio Pietrosanti (naif) <span dir="ltr"><<a href="mailto:lists@infosecurity.ch">lists@infosecurity.ch</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
Hi,<br>
<br>
now i start to better understand.<br>
<br>
I am not using STUN or ICE and in my case clients are in 2 situations:<br>
a) The client has a private ip address between a NAT Port address
transalation<br>
b) The client has a public ip address but that address is firewalled
(3G operator that provide a public IP address but does not let incoming
connection to go trough).<br>
<br>
In that scenario i got one-way audio with only one "Auto changing port
from ..." messages from FS debug log.<br>
<br>
Is there a way to force the FS not to look at the IP inside the SDP
header provisioned by the SIP UA, considering the IP address always
"buggy", so always not providing the right IP address configuration ?<br><font color="#888888">
<br>
Fabio</font><div><div></div><div class="h5"><br>
<br>
On 23/03/10 17.51, Anthony Minessale wrote:
<blockquote type="cite">That message is a symptom of the UA lying in it's sdp.<br>
it says one ip in the sdp but means another.<br>
<br>
We start by sending all the media to the ip specified in the SDP
because that is the only IP we know.<br>
Then there is a small window at the beginning of the call where we
assume a packet sent to us where we specified in our SDP may be a
packet from a mis-configured UA, so we will change our destination to
that ip:port to defeat the bug in the client.<br>
<br>
in order to engage this fix the client has to send us several packets
so we can prove it's correct to change our target towards them.<br>
<br>
<br>
<div class="gmail_quote">On Tue, Mar 23, 2010 at 11:39 AM, Fabio
Pietrosanti (naif) <span dir="ltr"><<a href="mailto:lists@infosecurity.ch" target="_blank">lists@infosecurity.ch</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I
am having a similar problem with SIP UA based on PJSIP.<br>
<br>
My main issue relate with sometimes having one-way audio between two
peers.<br>
Exactly the same peers sometimes works, sometime does not work.<br>
<br>
What i see is that when everything works are two statement in debug log:<br>
[INFO] switch_rtp.c:2005 Auto Changing port from ...<br>
[INFO] switch_rtp.c:2005 Auto Changing port from ...<br>
<br>
When things does not work (one-way audio) i just see 1 line:<br>
[INFO] switch_rtp.c:2005 Auto Changing port from ...<br>
<br>
For all calls i am using inbound-proxy-media=true .<br>
<br>
Is the same issue you see/encounter?<br>
<font color="#888888"><br>
Fabio<br>
</font>
<div><br>
On 23/03/10 17.26, Madovsky wrote:<br>
> thanks for your answer Jason.<br>
> yes the problem is yet there. before<br>
> to submit a bug I want to be sure it's a mistake I did.<br>
> so when I set inbound-bypass-media to true, the problem disappear.<br>
> it's for sure a media problem.<br>
> A phone with a public IP ( I tested on 6 PCs) registers well, but
make a rtp<br>
> timeout<br>
> A phone behind a firewall (tested on 6 PCs too) registers well and
doesn't<br>
> make rtp timeout.<br>
> I stopped all IP rules like iptables etc... without success.<br>
> I'd like to understand<br>
><br>
> Thanks<br>
><br>
<br>
<br>
</div>
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