Hello Support,
As a proxy server , How many concurrent calls can be processed by a
Kamailio which is installed on a VM with 4 cores and 16GB of RAM?
Your answer is much appreciated.
--
*Regards,*
*Abtin Mir*
Thank you Konstantin,
What I can’t understand is why media packets are sent correctly to the public IP , while early media (RBT) are sent on private.
Regards,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru>
Sent: Monday, July 23, 2018 1:38 PM
To: Ali Taher <ataher(a)vanrise.com>
Subject: Re[4]: [SR-Users] NATHELPER issue
Original INVITE contains private IP in c= parameter, so you need to use STUN or TURN or ICE, something which is supported in your system or in your softphone.
See my comments in the previous mail.
Best regrads
Понедельник, 23 июля 2018, 12:25 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >:
Hi Konstantin,
INVITE:
INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0
Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d>
X-AUTH-IP: Y.Y.Y.Y
Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0
Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Max-Forwards: 69
Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1>
To: "9613159370"<sip:9613159370@X.X.X.X:5065>
From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 434
v=0
o=- 5 2 IN IP4 192.168.26.3
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.26.3
t=0 0
m=audio 8602 RTP/AVP 0 8 101
a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602
a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602
a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602
a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF
200 OK
200 OK
Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d>
From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d
To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Contact: <sip:9613159370@X.X.X.X:5060;transport=udp>
User-Agent: FIKAR
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X <mailto:506251539613159370@X.X.X.X> >;party=calling;privacy=off;screen=no
v=0
o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X
s=FIKAR
c=IN IP4 X.X.X.X
t=0 0
m=audio 24496 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
where:
X.X.X.X is Kamailio IP
Y.Y.Y.Y is origination public IP
Regards,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> >
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >
Subject: Re[2]: [SR-Users] NATHELPER issue
Hello Ali,
Could you show your latest INVITE and OK?
Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy).
Kamailio doesn't process media as you know.
Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA.
Your media software or hardware should support STUN or ICE protocols to work correctly via NAT.
Best regards.
Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> >:
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP.
There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru <file://e.mail.ru/compose/%3fmailto=mailto%253apiligrim_pk@mail.ru> >
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users(a)lists.kamailio.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >
To: <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <mailto:070901d42001$939be2b0$bad3a810$@vanrise.com> >
Content-Type: text/plain; charset="utf-8"
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze
To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze>
Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <mailto:CbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX>
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX
;orig-ioi=MXXXXXXXXXXXXXX
Content-Type: application/sdp
Contact: <sip:172.16.45.65:5060;transport=UDP>
Content-Length: 268
v=0
o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX
s=-
c=IN IP4 172.16.45.144
t=0 0
a=sendrecv
m=audio 47588 RTP/AVP 18 96
c=IN IP4 172.16.45.144
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:20
Following is the header of the sent ACK packet:
Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0
Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze>
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0
Via: SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc
Max-Forwards: 69
From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze
To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000
Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <mailto:CbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX>
CSeq: 125698370 ACK
Content-Length: 0
Where X.X.X.X is Kamailio server public IP.
Following is part of my config file :
route {
route(NATDETECT);
record_route();
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
route(DLGURI);
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
route(DLGURI);
if(t_check_trans()) {
t_relay();
}
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
.....
}
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
add_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}
Can you please check why the ACK is still sent on private IP ?
Thanks
Ali Taher
"Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> >
С уважением,
Константин Поляков.
С уважением,
Константин Поляков.
Hello there!
Using kamailio 5.0.1 and msilo module for storing offline messages I
noticed that sometimes, especially when a big number of messages dumping
for one user, some of them can arrive to receiver in wrong order.
Lets say we have 20 messages and the order of receiving could be next
1..18..17..19..20.
That could be a reason of that ?
Thanks,
Philip
Hello, can somebody help me how to configure Kamailio to route/forward
requests to different (backup) server when Kamailio itself is not able
to deliver them?
I have working SIP gateway (on different host/port) which deliver
received SIP messages to XMPP network and I want to achieve:
Kamailio should try to send SIP messages directly to foreign SIP domains
and when it fails (e.g. user not available or foreign SIP is not
running), then forward message to that my SIP <--> XMPP gateway which
will try to deliver it via XMPP.
I guess that Kamailio should be able to configure for such custom
routing/forwarding but I have not figured yet how to do it. I'm using
Kamailio 4.4 with slightly modified default kamailio.cfg config.
--
Pali Rohár
pali.rohar(a)gmail.com
Hi Konstantin,
INVITE:
INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0
Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d>
X-AUTH-IP: Y.Y.Y.Y
Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0
Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Max-Forwards: 69
Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1>
To: "9613159370"<sip:9613159370@X.X.X.X:5065>
From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 434
v=0
o=- 5 2 IN IP4 192.168.26.3
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.26.3
t=0 0
m=audio 8602 RTP/AVP 0 8 101
a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602
a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602
a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602
a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF
200 OK
200 OK
Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d>
From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d
To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Contact: <sip:9613159370@X.X.X.X:5060;transport=udp>
User-Agent: FIKAR
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X>;party=calling;privacy=off;screen=no
v=0
o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X
s=FIKAR
c=IN IP4 X.X.X.X
t=0 0
m=audio 24496 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
where:
X.X.X.X is Kamailio IP
Y.Y.Y.Y is origination public IP
Regards,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru>
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher <ataher(a)vanrise.com>
Subject: Re[2]: [SR-Users] NATHELPER issue
Hello Ali,
Could you show your latest INVITE and OK?
Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy).
Kamailio doesn't process media as you know.
Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA.
Your media software or hardware should support STUN or ICE protocols to work correctly via NAT.
Best regards.
Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >:
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP.
There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> >
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> >
To: <sr-users(a)lists.sip-router.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.sip%252drouter.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253a070901d42001$939be2b0$bad3a810$@vanrise.com> >
Content-Type: text/plain; charset="utf-8"
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze
To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze>
Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX>
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX
;orig-ioi=MXXXXXXXXXXXXXX
Content-Type: application/sdp
Contact: <sip:172.16.45.65:5060;transport=UDP>
Content-Length: 268
v=0
o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX
s=-
c=IN IP4 172.16.45.144
t=0 0
a=sendrecv
m=audio 47588 RTP/AVP 18 96
c=IN IP4 172.16.45.144
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:20
Following is the header of the sent ACK packet:
Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0
Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze>
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0
Via: SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc
Max-Forwards: 69
From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze
To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000
Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX>
CSeq: 125698370 ACK
Content-Length: 0
Where X.X.X.X is Kamailio server public IP.
Following is part of my config file :
route {
route(NATDETECT);
record_route();
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
route(DLGURI);
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
route(DLGURI);
if(t_check_trans()) {
t_relay();
}
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
.....
}
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
add_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}
Can you please check why the ACK is still sent on private IP ?
Thanks
Ali Taher
"Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> >
С уважением,
Константин Поляков.
Hi Konstantin,
INVITE:
INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0
Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d>
X-AUTH-IP: Y.Y.Y.Y
Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0
Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Max-Forwards: 69
Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1>
To: "9613159370"<sip:9613159370@X.X.X.X:5065>
From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 434
v=0
o=- 5 2 IN IP4 192.168.26.3
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.26.3
t=0 0
m=audio 8602 RTP/AVP 0 8 101
a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602
a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602
a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602
a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF
200 OK
200 OK
Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d>
From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d
To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Contact: <sip:9613159370@X.X.X.X:5060;transport=udp>
User-Agent: FIKAR
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X>;party=calling;privacy=off;screen=no
v=0
o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X
s=FIKAR
c=IN IP4 X.X.X.X
t=0 0
m=audio 24496 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Regards,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru>
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher <ataher(a)vanrise.com>
Subject: Re[2]: [SR-Users] NATHELPER issue
Hello Ali,
Could you show your latest INVITE and OK?
Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy).
Kamailio doesn't process media as you know.
Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA.
Your media software or hardware should support STUN or ICE protocols to work correctly via NAT.
Best regards.
Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >:
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP.
There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> >
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> >
To: <sr-users(a)lists.sip-router.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.sip%252drouter.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253a070901d42001$939be2b0$bad3a810$@vanrise.com> >
Content-Type: text/plain; charset="utf-8"
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze
To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze>
Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX>
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX
;orig-ioi=MXXXXXXXXXXXXXX
Content-Type: application/sdp
Contact: <sip:172.16.45.65:5060;transport=UDP>
Content-Length: 268
v=0
o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX
s=-
c=IN IP4 172.16.45.144
t=0 0
a=sendrecv
m=audio 47588 RTP/AVP 18 96
c=IN IP4 172.16.45.144
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:20
Following is the header of the sent ACK packet:
Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0
Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze>
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0
Via: SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc
Max-Forwards: 69
From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze
To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000
Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX>
CSeq: 125698370 ACK
Content-Length: 0
Where X.X.X.X is Kamailio server public IP.
Following is part of my config file :
route {
route(NATDETECT);
record_route();
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
route(DLGURI);
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
route(DLGURI);
if(t_check_trans()) {
t_relay();
}
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
.....
}
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
add_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}
Can you please check why the ACK is still sent on private IP ?
Thanks
Ali Taher
"Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> >
С уважением,
Константин Поляков.
Hi,
I have a kamailio v5.0 setup where cdrs are enabled using acc module. Is
there a way to enable / disable cdrs on per call basis in kamailio script?
For example i want to allow cdrs only for offnet calls and disable them for
onnet calls?
May be a dialog flag configurable in module parameters to signal acc module
whether or not to generate cdr for current dialog at the end of call?
