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; };