[SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold
Sergiu Pojoga
pojogas at gmail.com
Fri Mar 23 13:50:26 CET 2018
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 at 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 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-bounces at lists.kamailio.org] *On Behalf
> Of *gerry kernan
> *Sent:* 23 March 2018 08:50
> *To:* 'Kamailio (SER) - Users Mailing List' <sr-users at 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 at lists.kamailio.org
> <sr-users-bounces at lists.kamailio.org>] *On Behalf Of *Sergiu Pojoga
> *Sent:* 23 March 2018 01:34
> *To:* Kamailio (SER) - Users Mailing List <sr-users at 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 at 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*
>
>
>
> [image: cid:image001.jpg at 01D105A5.2701B0E0]
>
>
>
> *Infinity IT | 17 The Mall | Beacon Court | Sandyford |
> Dublin D18 E3C8 | Ireland*
>
> *Tel: +353 - (0)1 - 293 0090 | E-Mail: *gerry.kernan at infinityit.ie
>
>
>
> *Managed IT Services Infinity IT* - www.infinityit.ie
>
> *IP Telephony Asterisk Consulting* –
> www.asteriskconsulting.com
>
> *Contact Centre Total Interact* – www.totalinteract.com
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
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