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-bounces@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.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 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@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] 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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