Re: [SR-Users] Re: Audio stops after resuming call from hold
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.ker...@infinityit.ie>
To: 'Kamailio (SER) - Users Mailing List' <sr-users@lists.kamailio.org>
Sent: 23/03/2018 3:55 PM
Subject: Re: [SR-Users] 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-boun...@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] 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.ker...@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-&
Re: [SR-Users] 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-boun...@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] 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.ker...@infinityit.ie
<mailto:gerry.ker...@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("a
Re: [SR-Users] Re: Audio stops after resuming call from hold
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-boun...@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] 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.ker...@infinityit.ie
<mailto:gerry.ker...@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_")) {
Re: [SR-Users] Re: Audio stops after resuming call from hold
}
}
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
$mb\n");
if(!is_method("ACK")) {
if (has_body("application/sdp")) {
xlog("L_NOTICE", "SDP OfferID=$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-boun...@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] 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-boun...@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: Re: [SR-Users] Audio stops after resuming call from hold
OMG, what are the odds, a client re
Re: [SR-Users] 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-boun...@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: 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.ker...@infinityit.ie <mailto:gerry.ker...@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.ker...@infinityit.ie> gerry.ker...@infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP TelephonyAsterisk Consulting – <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact CentreTotal 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 https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine nattede WAN address
Thanks Stefan That's sorted my issue Gerry Sent from my iPhone > On 1 Jun 2017, at 09:25, Mititelu Stefan <stefan.mititel...@gmail.com> wrote: > > Hi gerry, > > 1. The rtpengine is running on the same machine as kamailio? (From > --interface=192.X.X.148!X.X.X.X , I guess so..) > 2. Can you paste the rtpengine_offer/rtpengine_answer you are using in > Kamailio config? > > I remember trying this approach which worked for me: > - set interface = external/192.X.X.148!X.X.X.X;internal/192.X.X.148 > - use rtpengine_manage() in Kamailio config with "direction=external > direction=internal" for incoming calls > - use rtpengine_manage() in Kamailio config with "direction=internal > direction=external" for outgoing calls > > Give it a try! :) > > --- > Stefan > > >> On Wed, May 31, 2017 at 6:31 PM, gerry kernan <gerry.ker...@infinityit.ie> >> wrote: >> Hi >> >> >> >> I’m looking to check if what I’m trying to configure is feasible . >> >> >> >> UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN >> 192.X.X.131) >> >> >> >> I have a WN IP forwarded to and internal Kamailio server which is sitting in >> front of an asterisk server. SIP requests are sent to asterisk from Kamailio >> via dispatcher . >> >> SIP client can register via WAN and Kamailio to asterisk , I have rtpenine >> configured as below. >> >> >> >> /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 >> --timeout=60 --silent-timeout=3600 >> --pidfile=/var/run/ngcp-rtpengine-daemon.pid --table=0 --log-level=6 >> --log-facility=local7 --log-facility-rtcp=local7 --homer=192.168.200.5:9060 >> --homer-protocol=udp --homer-id=2002 >> >> >> >> >> >> But SIP SDP when sent to asterisk is still show the WAN IP of the Sip client >> in the SDP offer that goes to asterisk , ive tried adding parameter >> “via-branch=” and “auto-bridge” to rtpengine_offer but they don’t seem to >> make a difference , is there anyone way I can force the SDP offer that goes >> to the backend PBX to use the IP of Kamailio server that rtpengine is >> running on rather that the WAN IP that I’m advertising(X.X.X.X) ? >> >> >> >> >> >> >> >> Example SIP SDP from WAN client to asterisk >> >> >> >> INVITE sip:*9...@sip.domain.com;transport=UDP SIP/2.0 >> >> Via: SIP/2.0/UDP >> Client-IP:29345;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport >> Max-Forwards: 70 >> >> Contact: <sip:21@Client-IP:29345;transport=UDP> To: >> <sip:*9...@sip.domain.com;transport=UDP> >> >> From: <sip:2...@sip.domain.com;transport=UDP>;tag=e637343b >> >> Call-ID: jvqeIB9f48_OGcTq0UxP7w.. >> >> CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, >> OPTIONS, INFO, SUBSCRIBE >> >> Content-Type: application/sdp >> >> User-Agent: Zoiper rv2.8.40 >> >> o=Z >> >> 0 0 IN IP4 Client-IP >> >> s=Z >> >> c=IN IP4 Client-IP >> >> t=0 0 >> >> m=audio 29344 RTP/AVP 8 0 101 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=sendrecv >> >> >> >> and SDP that Kamailio sends to asterisk >> >> >> >> INVITE sip:*9...@sip.domain.com;transport=UDP SIP/2.0 >> >> Record-Route: <sip:192.X.X.148;lr> >> >> Via: SIP/2.0/UDP >> 192.X.X.148;branch=z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0 >> >> Via: SIP/2.0/UDP >> Client-IP:29345;received=Client-IP;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport=29345 >> Max-Forwards: 69 >> >> Contact: <sip:21@Client-IP:29345;transport=UDP> To: >> <sip:*9...@sip.domain.com;transport=UDP> >> >> From: <sip:2...@sip.domain.com;transport=UDP>;tag=e637343b >> >> Call-ID: jvqeIB9f48_OGcTq0UxP7w.. >> >> CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, >> OPTIONS, INFO, SUBSCRIBE >> >> Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 >> >> Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> >> >> v=0 >> >> o=Z 0 0 IN IP4 Client-IP >> >> s=Z c=IN IP4 X.X.X.X >> >> t=0 0 >> >> m=audio 32550 RTP/AVP 8 0 101 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fm
[SR-Users] rtpengine nattede WAN address
Hi I'm looking to check if what I'm trying to configure is feasible . UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN 192.X.X.131) I have a WN IP forwarded to and internal Kamailio server which is sitting in front of an asterisk server. SIP requests are sent to asterisk from Kamailio via dispatcher . SIP client can register via WAN and Kamailio to asterisk , I have rtpenine configured as below. /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 --timeout=60 --silent-timeout=3600 --pidfile=/var/run/ngcp-rtpengine-daemon.pid --table=0 --log-level=6 --log-facility=local7 --log-facility-rtcp=local7 --homer=192.168.200.5:9060 --homer-protocol=udp --homer-id=2002 But SIP SDP when sent to asterisk is still show the WAN IP of the Sip client in the SDP offer that goes to asterisk , ive tried adding parameter "via-branch=" and "auto-bridge" to rtpengine_offer but they don't seem to make a difference , is there anyone way I can force the SDP offer that goes to the backend PBX to use the IP of Kamailio server that rtpengine is running on rather that the WAN IP that I'm advertising(X.X.X.X) ? Example SIP SDP from WAN client to asterisk INVITE sip:*9...@sip.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP Client-IP:29345;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport Max-Forwards: 70 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*9...@sip.domain.com;transport=UDP> From: <sip:2...@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 Client-IP t=0 0 m=audio 29344 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv and SDP that Kamailio sends to asterisk INVITE sip:*9...@sip.domain.com;transport=UDP SIP/2.0 Record-Route: <sip:192.X.X.148;lr> Via: SIP/2.0/UDP 192.X.X.148;branch=z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0 Via: SIP/2.0/UDP Client-IP:29345;received=Client-IP;branch=z9hG4bK-524287-1---e38082af1f110ad 1;rport=29345 Max-Forwards: 69 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*9...@sip.domain.com;transport=UDP> From: <sip:2...@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> v=0 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 X.X.X.X t=0 0 m=audio 32550 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:32551 a=ice-ufrag:Dwd23Dem a=ice-pwd:9Itp6Cq7lkYJxMkeDeA1I2kkzw a=candidate:KZyzUR5lXJJpt1Et 1 UDP 2130706431 X.X.X.X 32550 typ host a=candidate:KZyzUR5lXJJpt1Et 2 UDP 2130706430 X.X.X.X 32551 typ host Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.ker...@infinityit.ie> gerry.ker...@infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP TelephonyAsterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact CentreTotal Interact - <http://www.totalinteract.com> www.totalinteract.com ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio rtpengine sdp
Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk VM's. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider I only get audio on stream from asterisk registered ext to external caller , no audio from external caller to the asterisk ext. I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs from rtpenegine show it as trying to send the rtp to the private address of the sip endpoint rather that its WAN address. I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.ker...@infinityit.ie> gerry.ker...@infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP TelephonyAsterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact CentreTotal Interact - <http://www.totalinteract.com> www.totalinteract.com ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
