Dear List,
At below config. i setted up OpenSER on Public IP and PSTN Gateway is again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL modems on both client side and note that they have SIP ALG, also i've completed port forwarding in each modem, so SIP2SIP call happens (voice traffic goes from UA to UA) without need of anything extra as STUN or RTP RELAY. But since those are very away from each other if a UA needs to call PSTN the voice traffic flows from UA to PSTN and that lowers QoS :(
I would like to activate MediaProxy again but just for one case which is PSTN, so whenever a UA needs to call PSTN the voice traffic flows like :
UA (ATA) >>> OpenSER + MediaProxy >>> PSTN
How can i setup this like above without touching SIP2SIP calls between UA's... or adding the lines beginning with # these to my existing config is enough ?
Thanks, Ozan Blotter
debug=3 fork=yes log_stderror=no check_via=no listen=212.XXX.104.XXX # This is OpenSER's Public IP port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo"
# fifo_db_url="mysql://openser:openserrw@localhost/openser"
loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" # loadmodule "/usr/local/lib/openser/modules/mediaproxy.so"
modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") # modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") # modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients")
route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; };
if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; };
if (method!="REGISTER") { record_route(); };
if (loose_route()) { route(1); break; };
if (uri!=myself) { route(1); break; };
if (uri==myself) {
if (method=="REGISTER") { route(2); break; };
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # This is PSTN Gateways's Public IP # use_media_proxy(); route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }; }
route[1] { if (!t_relay()) { sl_reply_error(); }; }
route[2] { if (!save("location")) { sl_reply_error(); }; }
Hi Ozan,
so, if I get it right, you have a PSTN gw locate far away from the UAs...and instead of letting the media traffic to do directly between GW and UAs, you want to insert in the middle a RTP relay (mediaproxy) in order to improve the QoS, right? If so, I would say this will not help you - having a media relay on your media path leads to QoS degradation due additional delays....
If you still want to go for it....I will take a look on tour script.
regards, bogdan
Ozan Blotter wrote:
Dear List,
At below config. i setted up OpenSER on Public IP and PSTN Gateway is again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL modems on both client side and note that they have SIP ALG, also i've completed port forwarding in each modem, so SIP2SIP call happens (voice traffic goes from UA to UA) without need of anything extra as STUN or RTP RELAY. But since those are very away from each other if a UA needs to call PSTN the voice traffic flows from UA to PSTN and that lowers QoS :(
I would like to activate MediaProxy again but just for one case which is PSTN, so whenever a UA needs to call PSTN the voice traffic flows like :
UA (ATA) >>> OpenSER + MediaProxy >>> PSTN
How can i setup this like above without touching SIP2SIP calls between UA's... or adding the lines beginning with # these to my existing config is enough ?
Thanks, Ozan Blotter
Also in terms of ping times how far is the PSTN gateway, as Bogdan said, media proxy will add even more delay, as for sending pstn via mediaproxy, you could do all yourprocessing, and once ser detects that call must goto PSTN, then do all the NAT checking ...this is a oversimplified view...
Iqbal
Bogdan-Andrei Iancu wrote:
Hi Ozan,
so, if I get it right, you have a PSTN gw locate far away from the UAs...and instead of letting the media traffic to do directly between GW and UAs, you want to insert in the middle a RTP relay (mediaproxy) in order to improve the QoS, right? If so, I would say this will not help you - having a media relay on your media path leads to QoS degradation due additional delays....
If you still want to go for it....I will take a look on tour script.
regards, bogdan
Ozan Blotter wrote:
Dear List,
At below config. i setted up OpenSER on Public IP and PSTN Gateway is again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL modems on both client side and note that they have SIP ALG, also i've completed port forwarding in each modem, so SIP2SIP call happens (voice traffic goes from UA to UA) without need of anything extra as STUN or RTP RELAY. But since those are very away from each other if a UA needs to call PSTN the voice traffic flows from UA to PSTN and that lowers QoS :(
I would like to activate MediaProxy again but just for one case which is PSTN, so whenever a UA needs to call PSTN the voice traffic flows like :
UA (ATA) >>> OpenSER + MediaProxy >>> PSTN
How can i setup this like above without touching SIP2SIP calls between UA's... or adding the lines beginning with # these to my existing config is enough ?
Thanks, Ozan Blotter
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
.