[SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold
Arik Halperin
arik.halperin at s3code.com
Mon Apr 16 15:19:27 CEST 2018
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 at 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 at lists.kamailio.org] On Behalf Of Sergiu Pojoga
> Sent: 23 March 2018 12:50
> To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> Subject: Re: [SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold
>
> Config code looks solid to me. Look at the 'c=' in SDP in the forward and reply re-INVITEs. If it gets properly overwritten (same way as it is for the dialog forming INVITE) when rtpengine is engaged, then I believe we are facing some kind of bug in the 4.2 version of Kamailio, something about this thread: https://lists.kamailio.org/pipermail/sr-users/2012-September/074567.html <https://lists.kamailio.org/pipermail/sr-users/2012-September/074567.html>
>
> I can't upgrade Kamailio at the moment to test my theory as it's a production environment, but may be you can?
>
> On Fri, Mar 23, 2018 at 6:17 AM, gerry kernan <gerry.kernan at infinityit.ie <mailto:gerry.kernan at infinityit.ie>> wrote:
>> Hi
>>
>> I think my issue is related to rtpengine when the call is take off hold. Im using a private address and a public address . below is route section of our Kamailio.cfg and do I have somethimg setup incorrectly for handleing re-invites?
>>
>>
>> /usr/sbin/rtpengine --pidfile /var/run/rtpengine.pid --table=-1 --interface=priv/192.X.X.X --interface=pub/212.X.X.X --listen-ng=127.0.0.1:7722 <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 at lists.kamailio.org <mailto:sr-users-bounces at lists.kamailio.org>] On Behalf Of gerry kernan
>> Sent: 23 March 2018 08:50
>> To: 'Kamailio (SER) - Users Mailing List' <sr-users at lists.kamailio.org <mailto:sr-users at lists.kamailio.org>>
>> Subject: Re: [SR-Users] <UNJUNKED> Re: Audio stops after resuming call from hold
>>
>> Hi Segriu
>>
>> I think my issue is with rtpengine . I’m using direction parameter to set a LAN and WAN IP on the offer and I think it’s getting messed up during re-invites
>>
>>
>>
>>
>>
>> Best Regards
>>
>> Gerry Kernan
>>
>> From: sr-users [mailto:sr-users-bounces at lists.kamailio.org <mailto:sr-users-bounces at lists.kamailio.org>] On Behalf Of Sergiu Pojoga
>> Sent: 23 March 2018 01:34
>> To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org <mailto:sr-users at lists.kamailio.org>>
>> Subject: <UNJUNKED> Re: [SR-Users] Audio stops after resuming call from hold
>>
>> OMG, what are the odds, a client reported the same problem today! Edge proxy running same 4.2.3, requests are forwarded to a farm of Asterisks v13 in a similar way based on $rd, far-end NAT traversal is handled by Kamailio.
>>
>> I've had only an hour or so to debug today. Re-invites containing SDP are handled the same way as invites in terms of SDP mangling, all looks good in that sense. There's nothing special to be done about re-invites.
>>
>> Preliminary clue is that this happens (or not) depending on the type of firewall/NAT behind which the phone is located. In the case with the trouble, it's a Sonicwall, probably a Symmetric NAT. Is doesn't happen to a phone behind a Full/Restricted Cone NAT.
>>
>> What nat= are you setting for Asterisk peers?
>> Do you engage rtpproxy/rtpengine?
>> Any far-end NAT traversal manipulations involved such as SIP ALG or STUN?
>>
>> Cheers.
>>
>> On Thu, Mar 22, 2018 at 3:55 PM, gerry kernan <gerry.kernan at infinityit.ie <mailto:gerry.kernan at infinityit.ie>> wrote:
>>> Hi
>>>
>>> Hoping someone can point me in the right direction.
>>> I have a Kamailio Ver: 4.2.3-1.1 running in front of a few asterisk servers Ver: 13.17.2 sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold after resuming the call the party that placed the call on hold can’t hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold?
>>>
>>>
>>> Gerry Kernan
>>>
>>> <image001.jpg>
>>>
>>> Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland
>>> Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan at infinityit.ie <mailto:gerry.kernan at 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 at lists.kamailio.org <mailto:sr-users at 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 at lists.kamailio.org <mailto:sr-users at 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 at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20180416/dea5a5de/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 3751 bytes
Desc: not available
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20180416/dea5a5de/attachment.bin>
More information about the sr-users
mailing list