25 Oct
2006
25 Oct
'06
10:14 a.m.
- Make sure nat=yes is found in the Route set of the reINVITE.
- Look at the mediaproxy messages in /var/log/messages, you should get
one for the INVITE and then one for the OK, but not '' empty response.
g-)
Shaun Hofer wrote:
Hi,
I've been having a problem, where audio is lost either in one or both
directions when conversaion is taken off 'hold'. The parties are both behind
NAT and depending UA as whether one or both loose audio. From what I can tell
its to do with my loose route and nathelper, and how my ser.cfg deals with
the take off hold INVITE from the phones. When the call is taken off hold the
rtp streams aren't setup properly again (eg not using mediaproxy correctly).
What is the best way to solve this problem?
I've seen similarly posts to the mailing list about this problem with no
solution posted.
http://lists.iptel.org/pipermail/serusers/2006-March/027424.html
http://lists.iptel.org/pipermail/serusers/2006-April/027885.html
http://lists.iptel.org/pipermail/serusers/2006-May/028407.html
Thanks
Shaun
I have a similarly config to getting started guides ser.cfg
# -----------------------------------------------------------------
# Loose Route Section
# -----------------------------------------------------------------
if (loose_route()) {
if (!has_totag()) {
sl_send_reply("403", "Forbidden");
break;
};
if (method=="INVITE") {
if ((method=="INVITE" || method=="REFER")
&& !has_totag()) {
if
(!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use
From=ID");
break;
};
consume_credentials();
};
if
(client_nat_test("3")||search("^Route:.*;nat=yes")) {
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
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