Re: [SR-Users] Re: Audio stops after resuming call from hold

[Arik Halperin]

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.ker...@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-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 
> <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 SI

Re: [SR-Users] Re: Audio stops after resuming call from hold

[gerry kernan]

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

[gerry kernan]

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

[gerry kernan]

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

[Sergiu Pojoga]

 {
>
> 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
> $mb\n");
>
>
>
> if(!is_method("ACK")) {
>
> if (has_body("application/sdp")) {
>
> xlog("L_NOTICE", "SDP OfferID=$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_ali

Re: [SR-Users] Re: Audio stops after resuming call from hold

[gerry kernan]

   }

}

 

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

[gerry kernan]

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 
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  > wrote:

Hi 

 

Hoping someone can point me in the right direction.

I have a Kamailio Ver: 4.2.3-1.1  running in front of a few asterisk servers 
Ver: 13.17.2  sip is routed to an asterisk server depending the domain name in 
the sip request, all working as expected . but if a call is put on hold  after 
resuming the call the party that placed the call on hold can’t hear any audio. 
The other party can hear . do I need to do anything special to handle 
re-invites for calls put on hold?

 

 

Gerry Kernan

 



 

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