[SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold
gerry kernan
gerry.kernan at infinityit.ie
Sat Mar 24 09:52:01 CET 2018
Update didn't make any difference, when back to investigate reinvites
Added if statement that sets rtpengie direction if is invite with to tag , all ok now
Gerry Kernan
InfinityIT
Suite 17 The Mall
Beacon Court,
Sandyford,
Dublin 18.
p: +35312930090
f: +35312930137
w: www.infinityit.ie
From: gerry kernan <gerry.kernan at infinityit.ie>
To: 'Kamailio (SER) - Users Mailing List' <sr-users at lists.kamailio.org>
Sent: 23/03/2018 3:55 PM
Subject: Re: [SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold
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 at lists.kamailio.org] On Behalf Of Sergiu Pojoga
Sent: 23 March 2018 12:50
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
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] 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
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
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