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<blockquote cite="mid172295055@web.de" type="cite">
<pre wrap="">finally I figured out how to change the sdp - part of the message by using force_rtpproxy.
All works fine, I can make calls in both directions, but i still got two problems:
1. If someone from outside(VoIP - Provider) calls my UA, i have a ~10 sec. delay before i hear audio.
- i checked my dumpfile, and saw that the delay is between the last 180 Ringing and the 200 OK.
(- However RTP - Packets are sent before the 200 OK, after a 183-Message. Dont understand why but I dont think it matters that much.)
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183 is early media probably sent by your UA. The call is not
established two-way until 200 OK. You don't say in which end you don't
hear audio...<br>
<blockquote cite="mid172295055@web.de" type="cite">
<pre wrap="">2. If the call is terminated from outside(VoIP - Provider) the BYE message is not handled correctly by SER.
- the previous messages from the Voip provider have all a via HF with the private IP of my UA. The BYE message doesnt have such a Via. All other Via and RR fields are present.
- also the BYE uri is somewaht strange. Instead of sip:123456@privateIPofUA it contains sip:123456@publicIPofSER.
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Look in the initial INVITE Record-Route set. BYE is a new request and
is created with Route headers copied from the Record-Route set
established as part of the establishment of the dialog. <br>
You don't provide enough info about what is happening to the BYE, just
your observations. Send a SIP trace.<br>
g-)<br>
<br>
<blockquote cite="mid172295055@web.de" type="cite">
<pre wrap="">Thanks in advance for any help Smile
Best regards
Sebastian.
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<blockquote type="cite">
<pre wrap="">-----Ursprüngliche Nachricht-----
Von: "Greger V. Teigre" <a class="moz-txt-link-rfc2396E" href="mailto:greger@teigre.com"><greger@teigre.com></a>
Gesendet: 10.08.06 15:41:31
An: Sebastian Gabris <a class="moz-txt-link-rfc2396E" href="mailto:Sebastian.Gabris@web.de"><Sebastian.Gabris@web.de></a>
CC: <a class="moz-txt-link-abbreviated" href="mailto:serusers@lists.iptel.org">serusers@lists.iptel.org</a>
Betreff: Re: [Serusers] Multihomed SER & NAT
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<pre wrap=""><!---->
fix_nated_contact() is meant for changing the IP address of the Contact
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<pre wrap="">header when SER has a public address and the UA has a private (i.e.
behind NAT). This means that it will rewrite Contact to contain the
source ip address:port found in the UDP/TCP header.
This is probably correct, because I assume the UA cannot be reached
from the outside anyway. If you record-route, subsequent requests
should go through your ser.
If you have problems with responses not being returned, have a look at
Via headers as they control how answers to i.e an INVITE are routed
back to the UA. You may want to have a look at the rr module double
record route parameter to make sure that SER creates one record-route
for the internal and one for the external interface.
g-)
Sebastian Gabris wrote:
After reading a bit more mails, I came to two possible problems:
1. fix_nated_contact() needs a registrar to work properly
I dont use a registrar, since i register at the VoIP - Provider.
or
2. fix_nated_contact() is not ment to be used on this scenario
Any ideas on this?
Sebastian
Hi everybody,
need some help on nattraversal.
i use a multihomed SER, one private IP & one public IP.
MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP)
If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address.
Instead o writing down his public IP, SER writes his private IP.
Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information.
After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :)
Here are the important bits of my config:
fork=yes
log_stderror=no mhomed=1
listen=10.x.x.x
listen=193.x.x.x
loadmodule "/usr/lib/ser/modules/nathelper.so"
modparam("usrloc", "db_mode", 0)
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
if (nat_uac_test("3")) {
if (method == "REGISTER" || ! search("^Record-Route:")) {
log(2,"LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
};
... and so on.
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</pre>
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<pre wrap=""><!---->
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