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.kernan@infinityit.ie gerry.kernan@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
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*
[image: cid:image001.jpg@01D105A5.2701B0E0]
*Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland*
*Tel: +353 - (0)1 - 293 0090 | E-Mail: *gerry.kernan@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 mailto: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
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland
Tel: +353 - (0)1 - 293 0090 | E-Mail: mailto:gerry.kernan@infinityit.ie gerry.kernan@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
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 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.kernan@infinityit.ie mailto: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
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland
Tel: +353 - (0)1 - 293 0090 | E-Mail: mailto:gerry.kernan@infinityit.ie gerry.kernan@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
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*
[image: cid:image001.jpg@01D105A5.2701B0E0]
*Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland*
*Tel: +353 - (0)1 - 293 0090 | E-Mail: *gerry.kernan@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
I’ve been testing with jitsi softphone from a different location( customer was using Zoiper which fails every time) and hold/unhold works every time, mightn’t be a Kamailio or rtpengine issue. I’ll do further tests to see if it local firewall/network
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 mailto: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 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-bounces@lists.kamailio.org 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 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-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 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.kernan@infinityit.ie mailto: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
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland
Tel: +353 - (0)1 - 293 0090 | E-Mail: mailto:gerry.kernan@infinityit.ie gerry.kernan@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
I had a similar issue with RTP engine. When I got hold and called rtpengine_manage it had errors.
I’m using rtpengine_manage, so doing something like this:
if(!is_present_hf("x-purpose")) { if(nat_uac_test("8")) { xlog("L_ERR","NATMANAGE DBG test 8\n");
if(ds_is_from_list()){ rtpengine_manage("replace-session-connection replace-origin direction=priv direction=pub"); } else { rtpengine_manage("replace-session-connection replace-origin direction=pub direction=priv"); } } else { if(ds_is_from_list()) { rtpengine_manage("replace-session-connection replace-origin trust-address direction=priv direction=pub"); } else { rtpengine_manage("replace-session-connection replace-origin trust-address direction=pub direction=priv"); } } }
The x-purpose is a header I added in my sip client whenever I do hold.
I hope that helps.
Best Regards, Arik
On 23 Mar 2018, at 16:50, gerry kernan gerry.kernan@infinityit.ie wrote:
I’ve been testing with jitsi softphone from a different location( customer was using Zoiper which fails every time) and hold/unhold works every time, mightn’t be a Kamailio or rtpengine issue. I’ll do further tests to see if it local firewall/network
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 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 mailto: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 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-bounces@lists.kamailio.org 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 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-bounces@lists.kamailio.org 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 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.kernan@infinityit.ie mailto: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
<image001.jpg>
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan@infinityit.ie mailto:gerry.kernan@infinityit.ie
Managed IT Services Infinity IT - www.infinityit.ie http://www.infinityit.ie/ IP Telephony Asterisk Consulting – www.asteriskconsulting.com http://www.asteriskconsulting.com/ Contact Centre Total Interact – www.totalinteract.com http://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 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 https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 mailto: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 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-bounces@lists.kamailio.org 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 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-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 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.kernan@infinityit.ie mailto: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
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland
Tel: +353 - (0)1 - 293 0090 | E-Mail: mailto:gerry.kernan@infinityit.ie gerry.kernan@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
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@infinityit.ie To: 'Kamailio (SER) - Users Mailing List' sr-users@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@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
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List 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 https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-
Arik Halperin -
gerry kernan -
Sergiu Pojoga