Hi Segriu
I’ve updated to 4.3. I’ll let you know how I go on with the new version Best Regards Gerry Kernan From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Sergiu Pojoga Sent: 23 March 2018 12:50 To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org> Subject: Re: [SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold Config code looks solid to me. Look at the 'c=' in SDP in the forward and reply re-INVITEs. If it gets properly overwritten (same way as it is for the dialog forming INVITE) when rtpengine is engaged, then I believe we are facing some kind of bug in the 4.2 version of Kamailio, something about this thread: https://lists.kamailio.org/pipermail/sr-users/2012-September/074567.html I can't upgrade Kamailio at the moment to test my theory as it's a production environment, but may be you can? On Fri, Mar 23, 2018 at 6:17 AM, gerry kernan <gerry.ker...@infinityit.ie <mailto:gerry.ker...@infinityit.ie> > wrote: Hi I think my issue is related to rtpengine when the call is take off hold. Im using a private address and a public address . below is route section of our Kamailio.cfg and do I have somethimg setup incorrectly for handleing re-invites? /usr/sbin/rtpengine --pidfile /var/run/rtpengine.pid --table=-1 --interface=priv/192.X.X.X --interface=pub/212.X.X.X --listen-ng=127.0.0.1:7722 <http://127.0.0.1:7722> --tos=184 --timeout=60 --log-level=7 --log-facility=local5 --homer-protocol=udp --homer-id=2011 request_route { route(SANITY); force_rport(); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) if (is_method("INVITE|SUBSCRIBE")) { record_route(); } if (af==INET) { route(SIPIPV4); } else { route(SIPIPV6); } } # Stateful fowarding route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if ( is_method("ACK") ) { # ACK is forwarded statelessly if (has_body("application/sdp")) { rtpengine_answer(); } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(DISPATCH); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } route[SIPIPV4] { if (src_ip != BACKEND_NET4) { # device (client) to server (backend) route(V4DEVTOSRV); } else { # server (backend) to devuce (client) route(V4SRVTODEV); } } route[SIPIPV6] { sl_send_reply("404", "Not routing for IPv6"); exit; } route[V4DEVTOSRV] { xlog("L_NOTICE", "client->backend FROM CLIENT IP: $si $rm $ru $td ID=$ci\n"); # SIP request packet client->backend # - remove preloaded route headers remove_hf("Route"); if (!lookup_domain("$td", "dattr_")) { xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " "found in domain table\n"); xlog("attempt to login with unkown domain from $si"); sl_send_reply("404", "No route for domain"); exit; } if (!defined $avp(dattr_routeset)) { xlog("L_ERR", "$si $rm $ru -- attribute \"routeset\" is " + "undefined for domain $td\n"); sl_send_reply("404", "No route id for domain"); exit; } if( !ds_select_dst(4000 + $avp(dattr_routeset), "1") ) { xlog("L_NOTICE", "Drop....\n"); sl_send_reply("404", "No destination"); } if (is_method("REGISTER")) { add_path_received(); } else { if (nat_uac_test("19")) { if(is_first_hop()) { add_contact_alias(); } } } if (has_body("application/sdp")) { rtpengine_offer("direction=pub direction=priv ICE=remove"); } route(DISPATCH); xlog("L_NOTICE", "DISPATCH: source address: $si SIP request's method: $rm SIP Request's URI: $ru ID=$ci\n"); exit; } route[V4SRVTODEV] { # SIP request packet backend->client # Invites from backend contain Route field and it should be used # to reach the registered client xlog("L_NOTICE", "backend->client FROM BACKEND: source address: $si" " METHOD: $rm $ru To-URI: $tu ID=$ci \n"); xlog("L_NOTICE", "backend->client $rm: TO $ru FROM $fu ID=$ci\n"); if (has_body("application/sdp")) { rtpengine_offer("direction=priv direction=pub ICE=remove"); } if(!is_present_hf("Route")) { sl_send_reply("404", "No record routing"); exit; } loose_route(); route(DISPATCH); } route[DISPATCH] { xlog("L_NOTICE", "ROUTE-DISPATCH $si $rm $ru ID=$ci \n"); xlog("L_NOTICE", "ROUTE-DISPATCH Messege buff.... ID=$ci $rm \n $mb\n"); if(!is_method("ACK")) { if (has_body("application/sdp")) { xlog("L_NOTICE", "SDP Offer....ID=$ci\n"); t_on_reply("INVSDP"); } else { t_on_reply("INVNOSDP"); } } xlog("L_NOTICE", "DISPATCH $si METHOD: $rm $ru $du ID=$ci\n"); xlog("L_NOTCIE", "Return code: $rc ID=$ci\n"); route(RELAY); exit; } # URI update for dialog requests route[DLGURI] { if(!isdsturiset()) { handle_ruri_alias(); } return; } route[REPLYALIAS] { if(src_ip != BACKEND_NET4) { # SIP reply packet client->backend xlog("L_NOTICE", "FROM CLIENT($si onreply_route- ): Method: $rm" "$ru To: $tu Recieved on: $Ri ID=$ci "); add_contact_alias(); } else { # SIP reply packet backend->client xlog("L_NOTICE", "FROM BACKEND($si onreply_route): Method: $rm" " $ru To: $tu Recieved on: $Ri ID=$ci"); xlog("L_NOTICE", "FROM BACKEND #rtpengine_answer# ($si onreply_route):" " source address: $si SIP request's method: $rm SIP Request's" " URI: $ru ID=$ci\n"); } } onreply_route[INVSDP] { if (af!=INET) { exit; } if (has_body("application/sdp")) { xlog("L_NOTICE", "INVSDP Route: Method: $rm" " $ru To: $tu Recieved on: $Ri ID=$ci\n $mb\n"); rtpengine_answer(); } route(REPLYALIAS); exit; } onreply_route[INVNOSDP] { if (af!=INET) { exit; } if (has_body("application/sdp")) { xlog("L_NOTICE", "INVNOSDP Route: Method: $rm" " $ru To: $tu Recieved on: $Ri ID=$ci\n $mb\n"); if(src_ip == BACKEND_NET4) { rtpengine_offer("direction=priv direction=pub ICE=remove"); } else { rtpengine_offer("direction=pub direction=priv ICE=remove"); } } route(REPLYALIAS); exit; } Best Regards Gerry Kernan From: sr-users [mailto:sr-users-boun...@lists.kamailio.org <mailto:sr-users-boun...@lists.kamailio.org> ] On Behalf Of gerry kernan Sent: 23 March 2018 08:50 To: 'Kamailio (SER) - Users Mailing List' <sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > Subject: Re: [SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold Hi Segriu I think my issue is with rtpengine . I’m using direction parameter to set a LAN and WAN IP on the offer and I think it’s getting messed up during re-invites Best Regards Gerry Kernan From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Sergiu Pojoga Sent: 23 March 2018 01:34 To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > Subject: <UNJUNKED> Re: [SR-Users] Audio stops after resuming call from hold OMG, what are the odds, a client reported the same problem today! Edge proxy running same 4.2.3, requests are forwarded to a farm of Asterisks v13 in a similar way based on $rd, far-end NAT traversal is handled by Kamailio. I've had only an hour or so to debug today. Re-invites containing SDP are handled the same way as invites in terms of SDP mangling, all looks good in that sense. There's nothing special to be done about re-invites. Preliminary clue is that this happens (or not) depending on the type of firewall/NAT behind which the phone is located. In the case with the trouble, it's a Sonicwall, probably a Symmetric NAT. Is doesn't happen to a phone behind a Full/Restricted Cone NAT. What nat= are you setting for Asterisk peers? Do you engage rtpproxy/rtpengine? Any far-end NAT traversal manipulations involved such as SIP ALG or STUN? Cheers. On Thu, Mar 22, 2018 at 3:55 PM, gerry kernan <gerry.ker...@infinityit.ie <mailto:gerry.ker...@infinityit.ie> > wrote: Hi Hoping someone can point me in the right direction. I have a Kamailio Ver: 4.2.3-1.1 running in front of a few asterisk servers Ver: 13.17.2 sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold after resuming the call the party that placed the call on hold can’t hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.ker...@infinityit.ie> gerry.ker...@infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting – <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact – <http://www.totalinteract.com> www.totalinteract.com _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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