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.kernan@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 --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 arerouted)
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
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 anddiscard
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 $tdID=$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'smethod: $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=pubICE=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# ($sionreply_route):"
" source address: $si SIP request'smethod: $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=pubICE=remove");
} else { rtpengine_offer("direction=pub direction=privICE=remove");
} } route(REPLYALIAS); exit;}
Best Regards
Gerry Kernan
*From:* sr-users [mailto:sr-users-bounces@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 *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-bounces@lists.kamailio.org sr-users-bounces@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 *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.kernan@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*
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