10 May
2021
10 May
'21
12:07 p.m.
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk)
---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when I
make a call to the number provided by the carrier and there is audio also
on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
10 May
10 May
12:24 p.m.
SDP is probably wrong, a trace here would help.
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
On Mon, May 10, 2021 at 10:11 AM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk)
---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when
I make a call to the number provided by the carrier and there is audio also
on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
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1:34 p.m.
Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.236.72.101:5060
;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk)
---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when
I make a call to the number provided by the carrier and there is audio also
on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
4:53 p.m.
Is the telecom operator on a private network? The 200 OK SDP is asking the
telco to send the rtp to 10.0.X.X.
The 200 OK (Kamailio->telco) the sdp says:
c=IN IP4 10.0.X.X
That should be an IP the telco can reach.
You need to configure kamailio and RTPProxy to set an IP the telco can
actually reach. And probably do it on both the INVITE and the 200 OK.
On the initial invite, you should do the the same.
On Mon, 10 May 2021 at 11:39, Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.236.72.101:5060
;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
trunk) ---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when
I make a call to the number provided by the carrier and there is audio also
on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
Edit mailing list options or unsubscribe:
* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
5:56 p.m.
Yes, the telecom operator is on the private network. 10.0.X.X is the SBC IP
of the telecom operator to which we register. 10.0.X.X is reachable only
through the second network interface. The complete flow is given below:
[image: image.png]
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network
interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is
telecom operator SBC IP to which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following
way:
*rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
udp:localhost:7722*
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.236.72.101:5060
;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
trunk) ---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when
I make a call to the number provided by the carrier and there is audio also
on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
11:49 p.m.
So let me get this right:
asterisk (10.0.x.x)--->(192.168.0.192) proxy (10.0.x.x)--->(10.0.x.x)telco
op
There's something i'm not seeing.
Can you explain further like i did above?
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
On Mon, May 10, 2021 at 8:42 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Yes, the telecom operator is on the private network.
10.0.X.X is the SBC
IP of the telecom operator to which we register. 10.0.X.X is reachable only
through the second network interface. The complete flow is given below:
[image: image.png]
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network
interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is
telecom operator SBC IP to which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following
way:
*rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
udp:localhost:7722*
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.236.72.101:5060
;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
Kamailio - Users
Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
Edit mailing list options or unsubscribe:
* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
trunk) ---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud
when I make a call to the number provided by the carrier and there is audio
also on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
__________________________________________________________ 
11 May
11 May
11:17 a.m.
Hello,
Could you share your Asterisk's sip context where you set up connectivity
with telco? Just delete all usernames and passwords, if any....
Your setup is very similar to the one I have with a very large telco in my
country, and it works fine as long as you always keep Asterisk on the RTP
path (no re-invites allowed, NAT always active)
*Sérgio Charrua*
*www.voip.pt <http://www.voip.pt/>*
Tel.: +351 <callto:+351+91+104+12+66>21 130 71 77
Email : *sergio.charrua(a)voip.pt <sergio.charrua(a)voip.pt>*
This message and any files or documents attached are strictly confidential
or otherwise legally protected.
It is intended only for the individual or entity named. If you are not the
named addressee or have received this email in error, please inform the
sender immediately, delete it from your system and do not copy or disclose
it or its contents or use it for any purpose. Please also note that
transmission cannot be guaranteed to be secure or error-free.
On Tue, May 11, 2021 at 8:02 AM David Villasmil <
david.villasmil.work(a)gmail.com> wrote:
So let me get this right:
asterisk (10.0.x.x)--->(192.168.0.192) proxy (10.0.x.x)--->(10.0.x.x)telco
op
There's something i'm not seeing.
Can you explain further like i did above?
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
On Mon, May 10, 2021 at 8:42 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
Yes, the telecom operator is on the private
network. 10.0.X.X is the SBC
IP of the telecom operator to which we register. 10.0.X.X is reachable only
through the second network interface. The complete flow is given below:
[image: image.png]
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first
network interface IP, 10.0.87.230 is second network interface IP
and 10.0.76.9 is telecom operator SBC IP to which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following
way:
*rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
udp:localhost:7722*
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
Edit mailing list options or unsubscribe:
* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Here are the SIP Traces:
*Asterisk Server to Kamailio Server (SDP Packet):*
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
*Kamailio Server to Telecom Operator Carrier (SDP Packet):*
2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 3.236.72.101:5060
;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X
:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
Kamailio - Users
Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
Edit mailing list options or unsubscribe:
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Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
trunk) ---> Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud
when I make a call to the number provided by the carrier and there is audio
also on both sides.
However, when I am making an outbound call from Asterisk server to the
number through Kamailio, there is no audio when I pick up the call. I have
tried to capture the traces but not able to understand the exact problem
here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
__________________________________________________________ 
11:04 a.m.
This is how it looks:
- Asterisk is running on Cloud having a public IP (3.236.X.X)
- Kamailio is running on a Physical server having 2 NIC ports.
- One of them is connected to the SIP trunk and this NIC port local
IP is 10.0.87.X. We register to the SBC server (10.0.76.X) of telecom
operator carrier through this port.
- Second NIC port is connected to ILL for internet connection having
local IP as 192.168.0.192 and public IP as 14.X.X.X
- To make an outbound call, Asterisk Server (3.236.X.X) sends the call
to Kamailio server on public IP (14.X.X.X) and in turn Kamailio server
sends the call to telecom operator SBC (10.0.76.X) through 10.0.87.X port.
Here is the actual flow:
(Asterisk)
(Kamailio Local ILL Port) (Kamailio Local SIP
Interface) (Telecom Operator SBC)
3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────
20:24:11.644416 │ INVITE (SDP) │
│ │
+0.000585 │ ──────────────────────────> │
│ │
20:24:11.645001 │ 100 trying -- your call is │
│ │
+0.000235 │ <────────────────────────── │
│ │
20:24:11.645236 │ │
│ INVITE (SDP) │
+0.005768 │ │
│ ──────────────────────────> │
20:24:11.651004 │ │
│ 100 Trying │
+0.580627 │ │
│ <────────────────────────── │
20:24:12.231631 │ │
│ 183 Session Progress (SDP) │
+0.000159 │ │
│ <────────────────────────── │
20:24:12.231790 │ 183 Session Progress (SDP) │
│ │
+1.932655 │ <────────────────────────── │
│ │
20:24:14.164445 │ │
│ 180 Ringing │
+0.000204 │ │
│ <────────────────────────── │
20:24:14.164649 │ 180 Ringing │
│ │
+3.631157 │ <────────────────────────── │
│ │
20:24:17.795806 │ │
│ 200 OK (SDP) │
+0.000361 │ │
│ <────────────────────────── │
20:24:17.796167 │ 200 OK (SDP) │
│ │
+0.233102 │ <────────────────────────── │
│ │
20:24:18.029269 │ ACK │
│ │
+0.000385 │ ──────────────────────────> │
│ │
20:24:18.029654 │ │
│ ACK │
+11.647190 │ │
│ ──────────────────────────> │
20:24:29.676844 │ │
│ BYE │
+0.000605 │ │
│ <────────────────────────── │
20:24:29.677449 │ BYE │
│ │
+0.236993 │ <────────────────────────── │
│ │
20:24:29.914442 │ 200 OK │
│ │
+0.000225 │ ──────────────────────────> │
│ │
20:24:29.914667 │ │
│ 200 OK │
│ │
│ ──────────────────────────> │
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 8:26 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Yes, the telecom operator is on the private network.
10.0.X.X is the SBC
IP of the telecom operator to which we register. 10.0.X.X is reachable only
through the second network interface. The complete flow is given below:
[image: image.png]
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network
interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is
telecom operator SBC IP to which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following
way:
*rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
udp:localhost:7722*
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
> Here are the SIP Traces:
>
> *Asterisk Server to Kamailio Server (SDP Packet):*
>
> 2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
>
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
> 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
> Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
> From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
> To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
> CSeq: 102 INVITE
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> Contact: <sip:09413745250@10.0.X.X
> :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> User-Agent: ZTE Softswitch/1.0.0
> Require: timer
> Session-Expires: 7200;refresher=uac
> Content-Length: 182
> Content-Type: application/sdp
>
> v=0
> o=- 1936 20890 IN IP4 10.0.X.X
> s=SBC call
> c=IN IP4 10.0.X.X
> t=0 0
> m=audio 37874 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:8 PCMA/8000/1
>
>
> *Kamailio Server to Telecom Operator Carrier (SDP Packet):*
>
> 2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 3.236.72.101:5060
> ;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
> Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
> From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
> To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
> CSeq: 102 INVITE
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> Contact: <sip:09413745250@10.0.X.X
> :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> User-Agent: ZTE Softswitch/1.0.0
> Require: timer
> Session-Expires: 7200;refresher=uac
> Content-Length: 182
> Content-Type: application/sdp
>
> v=0
> o=- 1936 20890 IN IP4 10.0.X.X
> s=SBC call
> c=IN IP4 10.0.X.X
> t=0 0
> m=audio 37874 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:8 PCMA/8000/1
>
> Regards
> Kashish
>
>
> On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
> kashishraheja1809(a)gmail.com> wrote:
>
>> Hi All,
>>
>> I have set up Kamailio in the following manner:
>>
>> Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
>> trunk) ---> Asterisk Server (on Cloud having public IP)
>>
>> I am successfully able to route the call to Asterisk server on Cloud
>> when I make a call to the number provided by the carrier and there is audio
>> also on both sides.
>>
>> However, when I am making an outbound call from Asterisk server to the
>> number through Kamailio, there is no audio when I pick up the call. I have
>> tried to capture the traces but not able to understand the exact problem
>> here.
>>
>> Note: I am running the RTP proxy on Kamailio server.
>>
>> Any help on why this might be happening?
>>
>> Thanks.
>> Regards
>> Kashish
>> +919413745250
>>
>
11:14 a.m.
This is how it looks:
- Asterisk is running on Cloud having a public IP (3.236.X.X)
- Kamailio is running on a Physical server having 2 NIC ports.
- One of them is connected to the SIP trunk and this NIC port local
IP is 10.0.87.X. We register to the SBC server (10.0.76.X) of telecom
operator carrier through this port.
- Second NIC port is connected to ILL for internet connection having
local IP as 192.168.0.192 and public IP as 14.X.X.X
- To make an outbound call, Asterisk Server (3.236.X.X) sends the call
to Kamailio server on public IP (14.X.X.X) and in turn Kamailio server
sends the call to telecom operator SBC (10.0.76.X) through 10.0.87.X port.
Here is the diagram:
3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────
20:24:11.644416 │ INVITE (SDP) │
│ │
+0.000585 │ ──────────────────────────> │
│ │
20:24:11.645001 │ 100 trying -- your call is │
│ │
+0.000235 │ <────────────────────────── │
│ │
20:24:11.645236 │ │
│ INVITE (SDP) │
+0.005768 │ │
│ ──────────────────────────> │
20:24:11.651004 │ │
│ 100 Trying │
+0.580627 │ │
│ <────────────────────────── │
20:24:12.231631 │ │
│ 183 Session Progress (SDP) │
+0.000159 │ │
│ <────────────────────────── │
20:24:12.231790 │ 183 Session Progress (SDP) │
│ │
+1.932655 │ <────────────────────────── │
│ │
20:24:14.164445 │ │
│ 180 Ringing │
+0.000204 │ │
│ <────────────────────────── │
20:24:14.164649 │ 180 Ringing │
│ │
+3.631157 │ <────────────────────────── │
│ │
20:24:17.795806 │ │
│ 200 OK (SDP) │
+0.000361 │ │
│ <────────────────────────── │
20:24:17.796167 │ 200 OK (SDP) │
│ │
+0.233102 │ <────────────────────────── │
│ │
20:24:18.029269 │ ACK │
│ │
+0.000385 │ ──────────────────────────> │
│ │
20:24:18.029654 │ │
│ ACK │
+11.647190 │ │
│ ──────────────────────────> │
20:24:29.676844 │ │
│ BYE │
+0.000605 │ │
│ <────────────────────────── │
20:24:29.677449 │ BYE │
│ │
+0.236993 │ <────────────────────────── │
│ │
20:24:29.914442 │ 200 OK │
│ │
+0.000225 │ ──────────────────────────> │
│ │
20:24:29.914667 │ │
│ 200 OK │
│ │
│ ──────────────────────────> │
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 8:26 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Yes, the telecom operator is on the private network.
10.0.X.X is the SBC
IP of the telecom operator to which we register. 10.0.X.X is reachable only
through the second network interface. The complete flow is given below:
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network
interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is
telecom operator SBC IP to which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following
way:
*rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
udp:localhost:7722*
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
> Here are the SIP Traces:
>
> *Asterisk Server to Kamailio Server (SDP Packet):*
>
> 2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
>
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
> 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
> Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
> From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
> To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
> CSeq: 102 INVITE
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> Contact: <sip:09413745250@10.0.X.X
> :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> User-Agent: ZTE Softswitch/1.0.0
> Require: timer
> Session-Expires: 7200;refresher=uac
> Content-Length: 182
> Content-Type: application/sdp
>
> v=0
> o=- 1936 20890 IN IP4 10.0.X.X
> s=SBC call
> c=IN IP4 10.0.X.X
> t=0 0
> m=audio 37874 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:8 PCMA/8000/1
>
>
> *Kamailio Server to Telecom Operator Carrier (SDP Packet):*
>
> 2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 3.236.72.101:5060
> ;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
> Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
> From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
> To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
> CSeq: 102 INVITE
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> Contact: <sip:09413745250@10.0.X.X
> :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> User-Agent: ZTE Softswitch/1.0.0
> Require: timer
> Session-Expires: 7200;refresher=uac
> Content-Length: 182
> Content-Type: application/sdp
>
> v=0
> o=- 1936 20890 IN IP4 10.0.X.X
> s=SBC call
> c=IN IP4 10.0.X.X
> t=0 0
> m=audio 37874 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:8 PCMA/8000/1
>
> Regards
> Kashish
>
>
> On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
> kashishraheja1809(a)gmail.com> wrote:
>
>> Hi All,
>>
>> I have set up Kamailio in the following manner:
>>
>> Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
>> trunk) ---> Asterisk Server (on Cloud having public IP)
>>
>> I am successfully able to route the call to Asterisk server on Cloud
>> when I make a call to the number provided by the carrier and there is audio
>> also on both sides.
>>
>> However, when I am making an outbound call from Asterisk server to the
>> number through Kamailio, there is no audio when I pick up the call. I have
>> tried to capture the traces but not able to understand the exact problem
>> here.
>>
>> Note: I am running the RTP proxy on Kamailio server.
>>
>> Any help on why this might be happening?
>>
>> Thanks.
>> Regards
>> Kashish
>> +919413745250
>>
>
11:25 a.m.
Since your asterisk server is in the cloud, make sure the relevant udp audio port range is
open to the telecoms carrier.
It is possible that your firewall/security group setup allows an incoming reply on a port
when something first goes out on that port which can explain why audio works in one
direction but not the other.
Blessings,
—
Daniel
On 11 May 2021, at 10:14, Kashish Raheja
<kashishraheja1809(a)gmail.com> wrote:
This is how it looks:
Asterisk is running on Cloud having a public IP (3.236.X.X)
Kamailio is running on a Physical server having 2 NIC ports.
One of them is connected to the SIP trunk and this NIC port local IP is 10.0.87.X. We
register to the SBC server (10.0.76.X) of telecom operator carrier through this port.
Second NIC port is connected to ILL for internet connection having local IP as
192.168.0.192 and public IP as 14.X.X.X
To make an outbound call, Asterisk Server (3.236.X.X) sends the call to Kamailio server
on public IP (14.X.X.X) and in turn Kamailio server sends the call to telecom operator SBC
(10.0.76.X) through 10.0.87.X port.
Here is the diagram:
3.236.72.101:5060 <http://3.236.72.101:5060/>
192.168.0.192:5060 <http://192.168.0.192:5060/> 10.0.87.230:5060
<http://10.0.87.230:5060/> 10.0.76.9:5060
<http://10.0.76.9:5060/>
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────
20:24:11.644416 │ INVITE (SDP) │ │
│
+0.000585 │ ──────────────────────────> │ │
│
20:24:11.645001 │ 100 trying -- your call is │ │
│
+0.000235 │ <────────────────────────── │ │
│
20:24:11.645236 │ │ │
INVITE (SDP) │
+0.005768 │ │ │
──────────────────────────> │
20:24:11.651004 │ │ │
100 Trying │
+0.580627 │ │ │
<────────────────────────── │
20:24:12.231631 │ │ │ 183
Session Progress (SDP) │
+0.000159 │ │ │
<────────────────────────── │
20:24:12.231790 │ 183 Session Progress (SDP) │ │
│
+1.932655 │ <────────────────────────── │ │
│
20:24:14.164445 │ │ │
180 Ringing │
+0.000204 │ │ │
<────────────────────────── │
20:24:14.164649 │ 180 Ringing │ │
│
+3.631157 │ <────────────────────────── │ │
│
20:24:17.795806 │ │ │
200 OK (SDP) │
+0.000361 │ │ │
<────────────────────────── │
20:24:17.796167 │ 200 OK (SDP) │ │
│
+0.233102 │ <────────────────────────── │ │
│
20:24:18.029269 │ ACK │ │
│
+0.000385 │ ──────────────────────────> │ │
│
20:24:18.029654 │ │ │
ACK │
+11.647190 │ │ │
──────────────────────────> │
20:24:29.676844 │ │ │
BYE │
+0.000605 │ │ │
<────────────────────────── │
20:24:29.677449 │ BYE │ │
│
+0.236993 │ <────────────────────────── │ │
│
20:24:29.914442 │ 200 OK │ │
│
+0.000225 │ ──────────────────────────> │ │
│
20:24:29.914667 │ │ │
200 OK │
│ │ │
──────────────────────────> │
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 8:26 PM Kashish Raheja <kashishraheja1809(a)gmail.com
<mailto:kashishraheja1809@gmail.com>> wrote:
Yes, the telecom operator is on the private network. 10.0.X.X is the SBC IP of the
telecom operator to which we register. 10.0.X.X is reachable only through the second
network interface. The complete flow is given below:
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network interface IP,
10.0.87.230 is second network interface IP and 10.0.76.9 is telecom operator SBC IP to
which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following way:
rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s udp:localhost:7722
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <kashishraheja1809(a)gmail.com
<mailto:kashishraheja1809@gmail.com>> wrote:
Here are the SIP Traces:
Asterisk Server to Kamailio Server (SDP Packet):
2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
<mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110>
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060
<http://sip:09413745250@192.168.0.192:5060/>>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Kamailio Server to Telecom Operator Carrier (SDP Packet):
2021/05/10 15:54:52.835419 192.168.0.192:5060 <http://192.168.0.192:5060/> ->
3.X.X.X:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
<mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110>
From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
To: <sip:09413745250@192.168.0.192:5060
<http://sip:09413745250@192.168.0.192:5060/>>;tag=aa2c806-Huku2c07186a1
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Contact: <sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
User-Agent: ZTE Softswitch/1.0.0
Require: timer
Session-Expires: 7200;refresher=uac
Content-Length: 182
Content-Type: application/sdp
v=0
o=- 1936 20890 IN IP4 10.0.X.X
s=SBC call
c=IN IP4 10.0.X.X
t=0 0
m=audio 37874 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000/1
Regards
Kashish
On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <kashishraheja1809(a)gmail.com
<mailto:kashishraheja1809@gmail.com>> wrote:
Hi All,
I have set up Kamailio in the following manner:
Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) --->
Asterisk Server (on Cloud having public IP)
I am successfully able to route the call to Asterisk server on Cloud when I make a call
to the number provided by the carrier and there is audio also on both sides.
However, when I am making an outbound call from Asterisk server to the number through
Kamailio, there is no audio when I pick up the call. I have tried to capture the traces
but not able to understand the exact problem here.
Note: I am running the RTP proxy on Kamailio server.
Any help on why this might be happening?
Thanks.
Regards
Kashish
+919413745250
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12:39 p.m.
You need to also
INVITE
Asterisk->kamailio offer the asterisk’s public IP in the SDP.
Kamailio->telco offer in the invite the rtpproxy’s prívate IP (10.x.x.x)
And on the 200
Kamailio->asterisk offer the rtpproxy’s PUBLIC ip.
On the cloud firewall, you need to allow the rtpproxy and Kamailio’s public
IPs.
On the asterisk, you need to configure its public IP as the SDP, not the
private.
You also need to start rtpproxy in bridge mode providing both the public
and private IPs at start (or config)
Hope this helps.
On Tue, 11 May 2021 at 09:32, Daniel Donoghue <daniel.donoghue(a)freespee.com>
wrote:
Since your asterisk server is in the cloud, make sure
the relevant udp
audio port range is open to the telecoms carrier.
It is possible that your firewall/security group setup allows an incoming
reply on a port when something first goes out on that port which can
explain why audio works in one direction but not the other.
Blessings,
—
Daniel
On 11 May 2021, at 10:14, Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
This is how it looks:
- Asterisk is running on Cloud having a public IP (3.236.X.X)
- Kamailio is running on a Physical server having 2 NIC ports.
- One of them is connected to the SIP trunk and this NIC port local
IP is 10.0.87.X. We register to the SBC server (10.0.76.X) of telecom
operator carrier through this port.
- Second NIC port is connected to ILL for internet connection
having local IP as 192.168.0.192 and public IP as 14.X.X.X
- To make an outbound call, Asterisk Server (3.236.X.X) sends the call
to Kamailio server on public IP (14.X.X.X) and in turn Kamailio server
sends the call to telecom operator SBC (10.0.76.X) through 10.0.87.X port.
Here is the diagram:
3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────
20:24:11.644416 │ INVITE (SDP) │ │
│
+0.000585 │ ──────────────────────────> │ │
│
20:24:11.645001 │ 100 trying -- your call is │ │
│
+0.000235 │ <────────────────────────── │ │
│
20:24:11.645236 │ │ │
INVITE (SDP) │
+0.005768 │ │ │
──────────────────────────> │
20:24:11.651004 │ │ │
100 Trying │
+0.580627 │ │ │
<────────────────────────── │
20:24:12.231631 │ │ │ 183
Session Progress (SDP) │
+0.000159 │ │ │
<────────────────────────── │
20:24:12.231790 │ 183 Session Progress (SDP) │ │
│
+1.932655 │ <────────────────────────── │ │
│
20:24:14.164445 │ │ │
180 Ringing │
+0.000204 │ │ │
<────────────────────────── │
20:24:14.164649 │ 180 Ringing │ │
│
+3.631157 │ <────────────────────────── │ │
│
20:24:17.795806 │ │ │
200 OK (SDP) │
+0.000361 │ │ │
<────────────────────────── │
20:24:17.796167 │ 200 OK (SDP) │ │
│
+0.233102 │ <────────────────────────── │ │
│
20:24:18.029269 │ ACK │ │
│
+0.000385 │ ──────────────────────────> │ │
│
20:24:18.029654 │ │ │
ACK │
+11.647190 │ │ │
──────────────────────────> │
20:24:29.676844 │ │ │
BYE │
+0.000605 │ │ │
<────────────────────────── │
20:24:29.677449 │ BYE │ │
│
+0.236993 │ <────────────────────────── │ │
│
20:24:29.914442 │ 200 OK │ │
│
+0.000225 │ ──────────────────────────> │ │
│
20:24:29.914667 │ │ │
200 OK │
│ │ │
──────────────────────────> │
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 8:26 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Yes, the telecom operator is on the private
network. 10.0.X.X is the SBC
IP of the telecom operator to which we register. 10.0.X.X is reachable only
through the second network interface. The complete flow is given below:
Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first
network interface IP, 10.0.87.230 is second network interface IP
and 10.0.76.9 is telecom operator SBC IP to which we do SIP register.
We are running the rtpproxy on local IP (192.168.0.192) in the following
way:
*rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
udp:localhost:7722*
Thanks.
Regards
Kashish
On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
> Here are the SIP Traces:
>
> *Asterisk Server to Kamailio Server (SDP Packet):*
>
> 2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
>
192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
> 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
> Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
> From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
> To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
> CSeq: 102 INVITE
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> Contact: <
> sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> User-Agent: ZTE Softswitch/1.0.0
> Require: timer
> Session-Expires: 7200;refresher=uac
> Content-Length: 182
> Content-Type: application/sdp
>
> v=0
> o=- 1936 20890 IN IP4 10.0.X.X
> s=SBC call
> c=IN IP4 10.0.X.X
> t=0 0
> m=audio 37874 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:8 PCMA/8000/1
>
>
> *Kamailio Server to Telecom Operator Carrier (SDP Packet):*
>
> 2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 3.236.72.101:5060
> ;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
> Call-ID: 58eb00885daef7ff3a67ad0e235e817a(a)14.98.22.110
> From: <sip:68XXXXX@10.0.X.X>;tag=as2b21d944
> To: <sip:09413745250@192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
> CSeq: 102 INVITE
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> Contact: <
> sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> User-Agent: ZTE Softswitch/1.0.0
> Require: timer
> Session-Expires: 7200;refresher=uac
> Content-Length: 182
> Content-Type: application/sdp
>
> v=0
> o=- 1936 20890 IN IP4 10.0.X.X
> s=SBC call
> c=IN IP4 10.0.X.X
> t=0 0
> m=audio 37874 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:8 PCMA/8000/1
>
> Regards
> Kashish
>
>
> On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
> kashishraheja1809(a)gmail.com> wrote:
>
>> Hi All,
>>
>> I have set up Kamailio in the following manner:
>>
>> Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
>> trunk) ---> Asterisk Server (on Cloud having public IP)
>>
>> I am successfully able to route the call to Asterisk server on Cloud
>> when I make a call to the number provided by the carrier and there is audio
>> also on both sides.
>>
>> However, when I am making an outbound call from Asterisk server to the
>> number through Kamailio, there is no audio when I pick up the call. I have
>> tried to capture the traces but not able to understand the exact problem
>> here.
>>
>> Note: I am running the RTP proxy on Kamailio server.
>>
>> Any help on why this might be happening?
>>
>> Thanks.
>> Regards
>> Kashish
>> +919413745250
>>
> __________________________________________________________
Kamailio - Users
Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
Edit mailing list options or unsubscribe:
* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
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20 May
20 May
11:14 p.m.
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't
seem to work. No audio in the outbound call however incoming call works
fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE
sip:09413745250@192.168.0.192:5060 SIP/2.0
3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;rport
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Max-Forwards: 70
01:22:15.782149 │ *INVITE (SDP) * │
│ │ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
+0.050579 │ *──────────────────────────>* │
│ │ │To:
<sip:09413745250@192.168.0.192:5060
01:22:15.832728 │ 100 trying -- your call is │
│ │ │Contact:
<sip:68983619@3.236.72.101:5060
+0.000348 │ <────────────────────────── │
│ │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
01:22:15.833076 │ │
│ INVITE (SDP) │ │CSeq: 102 INVITE
+0.004863 │ │
│ ──────────────────────────> │ │User-Agent: Asterisk
PBX 17.7.0
01:22:15.837939 │ │
│ 100 Trying │ │Date: Thu, 20 May
2021 19:52:15 GMT
+0.799120 │ │
│ <────────────────────────── │ │Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637059 │ │
│ 183 Session Progress (SDP) │ │Supported: replaces,
timer
+0.000179 │ │
│ <────────────────────────── │
│P-Preferred-Identity: <sip:68983600@10.0.76.9
01:22:16.637238 │ 183 Session Progress (SDP) │
│ │ │Content-Type:
application/sdp
+1.490537 │ <────────────────────────── │
│ │ │Content-Length: 263
01:22:18.127775 │ │
│ 180 Ringing │ │
+0.000189 │ │
│ <────────────────────────── │ │v=0
01:22:18.127964 │ 180 Ringing │
│ │ │o=root 1560151942
1560151942 IN IP4 3.236.72.101 *(Asterisk's Public IP)*
+0.349351 │ <────────────────────────── │
│ │ │s=Asterisk PBX
17.7.0
01:22:18.477315 │ │
│ 180 Ringing │ │c=IN IP4
3.236.72.101 *(Asterisk's Public IP)*
+0.000206 │ │
│ <<<──────────────────────── │ │t=0 0
01:22:18.477521 │ 180 Ringing │
│ │ │m=audio 14046
RTP/AVP 8 0 101
+19.181387 │ <<<──────────────────────── │
│ │ │a=rtpmap:8 PCMA/8000
01:22:37.658908 │ │
│ 200 OK (SDP) │ │a=rtpmap:0 PCMU/8000
+0.073719 │ │
│ <────────────────────────── │ │a=rtpmap:101
telephone-event/8000
01:22:37.732627 │ 200 OK (SDP) │
│ │ │a=fmtp:101 0-16
+0.241852 │ <────────────────────────── │
│ │ │a=maxptime:150
01:22:37.974479 │ ACK │
│ │ │a=sendrecv
+0.000282 │ ──────────────────────────> │
│ │ │
01:22:37.974761 │ │
│ ACK │ │
+4.095171 │ │
│ ──────────────────────────> │ │
01:22:42.069932 │ │
│ BYE │ │
+0.000361 │ │
│ <────────────────────────── │ │
01:22:42.070293 │ BYE │
│ │ │
+0.244125 │ <────────────────────────── │
│ │ │
01:22:42.314418 │ 200 OK │
│ │ │
+0.000275 │ ──────────────────────────> │
│ │ │
01:22:42.314693 │ │
│ 200 OK │ │
│ │
│ ──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE
sip:09413745250@10.0.76.9 SIP/2.0
3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Record-Route:
<sip:192.168.0.192;lr=on;ftag=as69eb1cce
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0
01:22:15.782149 │ INVITE (SDP) │
│ │ │Via: SIP/2.0/UDP
3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060
+0.050579 │ ──────────────────────────> │
│ │ │Max-Forwards: 69
01:22:15.832728 │ 100 trying -- your call is │
│ │ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
+0.000348 │ <────────────────────────── │
│ │ │To:
<sip:09413745250@192.168.0.192:5060
01:22:15.833076 │ │
│ * INVITE (SDP) * │ │Contact:
<sip:68983619@3.236.72.101:5060
+0.004863 │ │
│ *──────────────────────────>* │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
01:22:15.837939 │ │
│ 100 Trying │ │CSeq: 102 INVITE
+0.799120 │ │
│ <────────────────────────── │ │User-Agent: Asterisk
PBX 17.7.0
01:22:16.637059 │ │
│ 183 Session Progress (SDP) │ │Date: Thu, 20 May
2021 19:52:15 GMT
+0.000179 │ │
│ <────────────────────────── │ │Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637238 │ 183 Session Progress (SDP) │
│ │ │Supported: replaces,
timer
+1.490537 │ <────────────────────────── │
│ │
│P-Preferred-Identity: <sip:68983600@10.0.76.9
01:22:18.127775 │ │
│ 180 Ringing │ │Content-Type:
application/sdp
+0.000189 │ │
│ <────────────────────────── │ │Content-Length: 279
01:22:18.127964 │ 180 Ringing │
│ │ │
+0.349351 │ <────────────────────────── │
│ │ │v=0
01:22:18.477315 │ │
│ 180 Ringing │ │o=root 1560151942
1560151942 IN IP4 10.0.87.230 *(RTP Proxy's private IP)*
+0.000206 │ │
│ <<<──────────────────────── │ │s=Asterisk PBX
17.7.0
01:22:18.477521 │ 180 Ringing │
│ │ │c=IN IP4 10.0.87.230
*(RTP Proxy's private IP)*
+19.181387 │ <<<──────────────────────── │
│ │ │t=0 0
01:22:37.658908 │ │
│ 200 OK (SDP) │ │m=audio 37322
RTP/AVP 8 0 101
+0.073719 │ │
│ <────────────────────────── │ │a=rtpmap:8 PCMA/8000
01:22:37.732627 │ 200 OK (SDP) │
│ │ │a=rtpmap:0 PCMU/8000
+0.241852 │ <────────────────────────── │
│ │ │a=rtpmap:101
telephone-event/8000
01:22:37.974479 │ ACK │
│ │ │a=fmtp:101 0-16
+0.000282 │ ──────────────────────────> │
│ │ │a=maxptime:150
01:22:37.974761 │ │
│ ACK │ │a=sendrecv
+4.095171 │ │
│ ──────────────────────────> │ │a=nortpproxy:yes
01:22:42.069932 │ │
│ BYE │ │
+0.000361 │ │
│ <────────────────────────── │ │
01:22:42.070293 │ BYE │
│ │ │
+0.244125 │ <────────────────────────── │
│ │ │
01:22:42.314418 │ 200 OK │
│ │ │
+0.000275 │ ──────────────────────────> │
│ │ │
01:22:42.314693 │ │
│ 200 OK │ │
│ │
│ ──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK
3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Record-Route:
<sip:192.168.0.192;lr;ftag=as69eb1cce
01:22:15.782149 │ INVITE (SDP) │
│ │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
+0.050579 │ ──────────────────────────> │
│ │ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
01:22:15.832728 │ 100 trying -- your call is │
│ │ │To:
<sip:09413745250@192.168.0.192:5060>;tag=aa2c806-t7ln3f58c3ea1
+0.000348 │ <────────────────────────── │
│ │ │CSeq: 102 INVITE
01:22:15.833076 │ │
│ INVITE (SDP) │ │Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
+0.004863 │ │
│ ──────────────────────────> │ │Contact:
<sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff
01:22:15.837939 │ │
│ 100 Trying │ │User-Agent: ZTE
Softswitch/1.0.0
+0.799120 │ │
│ <────────────────────────── │ │Require: timer
01:22:16.637059 │ │
│ 183 Session Progress (SDP) │ │Session-Expires:
7200;refresher=uac
+0.000179 │ │
│ <────────────────────────── │ │Content-Length: 208
01:22:16.637238 │ 183 Session Progress (SDP) │
│ │ │Content-Type:
application/sdp
+1.490537 │ <────────────────────────── │
│ │ │
01:22:18.127775 │ │
│ 180 Ringing │ │v=0
+0.000189 │ │
│ <────────────────────────── │ │o=- 1026 13186 IN
IP4 192.168.0.192 *(RTP Proxy's public IP)*
01:22:18.127964 │ 180 Ringing │
│ │ │s=SBC call
+0.349351 │ <────────────────────────── │
│ │ │c=IN IP4
192.168.0.192 *(RTP Proxy's public IP)*
01:22:18.477315 │ │
│ 180 Ringing │ │t=0 0
+0.000206 │ │
│ <<<──────────────────────── │ │m=audio 48462
RTP/AVP 8 101
01:22:18.477521 │ 180 Ringing │
│ │ │a=rtpmap:101
telephone-event/8000
+19.181387 │ <<<──────────────────────── │
│ │ │a=fmtp:101 0-15
01:22:37.658908 │ │
│ 200 OK (SDP) │ │a=rtpmap:8
PCMA/8000/1
+0.073719 │ │
│ <────────────────────────── │ │a=nortpproxy:yes
01:22:37.732627 │ * 200 OK (SDP)* │
│ │ │
+0.241852 │ *<──────────────────────────* │
│ │ │
01:22:37.974479 │ ACK │
│ │ │
+0.000282 │ ──────────────────────────> │
│ │ │
01:22:37.974761 │ │
│ ACK │ │
+4.095171 │ │
│ ──────────────────────────> │ │
01:22:42.069932 │ │
│ BYE │ │
+0.000361 │ │
│ <────────────────────────── │ │
01:22:42.070293 │ BYE │
│ │ │
+0.244125 │ <────────────────────────── │
│ │ │
01:22:42.314418 │ 200 OK │
│ │ │
+0.000275 │ ──────────────────────────> │
│ │ │
01:22:42.314693 │ │
│ 200 OK │ │
│ │
│ ──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command:
*/usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -u
asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230
<http://192.168.0.192/10.0.87.230>*
Apart from this, in the Asterisk console I can see that the RTP packets are
being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts
029280, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts
029440, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts
029600, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts
029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for
incoming calls
Anything am I missing here?
Thanks.
Regards
Kashish
27 May
27 May
4:14 p.m.
Haven't been able to sort this out yet. Anything am I missing here?
Thanks.
Regards
Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't
seem to work. No audio in the outbound call however incoming call works
fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060
10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;rport
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Max-Forwards: 70
01:22:15.782149 │ *INVITE (SDP) * │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
+0.050579 │ *──────────────────────────>* │ │
│ │To: <sip:09413745250@192.168.0.192:5060>
01:22:15.832728 │ 100 trying -- your call is │ │
│ │Contact: <sip:68983619@3.236.72.101:5060>
+0.000348 │ <────────────────────────── │ │
│ │Call-ID: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
01:22:15.833076 │ │ │
INVITE (SDP) │ │CSeq: 102 INVITE
+0.004863 │ │ │
──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0
01:22:15.837939 │ │ │
100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT
+0.799120 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637059 │ │ │ 183
Session Progress (SDP) │ │Supported: replaces, timer
+0.000179 │ │ │
<────────────────────────── │ │P-Preferred-Identity:
<sip:68983600@10.0.76.9>
01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
+1.490537 │ <────────────────────────── │ │
│ │Content-Length: 263
01:22:18.127775 │ │ │
180 Ringing │ │
+0.000189 │ │ │
<────────────────────────── │ │v=0
01:22:18.127964 │ 180 Ringing │ │
│ │o=root 1560151942 1560151942 IN IP4 3.236.72.101
*(Asterisk's Public IP)*
+0.349351 │ <────────────────────────── │ │
│ │s=Asterisk PBX 17.7.0
01:22:18.477315 │ │ │
180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)*
+0.000206 │ │ │
<<<──────────────────────── │ │t=0 0
01:22:18.477521 │ 180 Ringing │ │
│ │m=audio 14046 RTP/AVP 8 0 101
+19.181387 │ <<<──────────────────────── │
│ │ │a=rtpmap:8 PCMA/8000
01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:0 PCMU/8000
+0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:101 telephone-event/8000
01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=fmtp:101 0-16
+0.241852 │ <────────────────────────── │ │
│ │a=maxptime:150
01:22:37.974479 │ ACK │ │
│ │a=sendrecv
+0.000282 │ ──────────────────────────> │ │
│ │
01:22:37.974761 │ │ │
ACK │ │
+4.095171 │ │ │
──────────────────────────> │ │
01:22:42.069932 │ │ │
BYE │ │
+0.000361 │ │ │
<────────────────────────── │ │
01:22:42.070293 │ BYE │ │
│ │
+0.244125 │ <────────────────────────── │ │
│ │
01:22:42.314418 │ 200 OK │ │
│ │
+0.000275 │ ──────────────────────────> │ │
│ │
01:22:42.314693 │ │ │
200 OK │ │
│ │ │
──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060
10.0.76.9:5060 │Record-Route:
<sip:192.168.0.192;lr=on;ftag=as69eb1cce>
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0
01:22:15.782149 │ INVITE (SDP) │ │
│ │Via: SIP/2.0/UDP
3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060
+0.050579 │ ──────────────────────────> │ │
│ │Max-Forwards: 69
01:22:15.832728 │ 100 trying -- your call is │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
+0.000348 │ <────────────────────────── │ │
│ │To: <sip:09413745250@192.168.0.192:5060>
01:22:15.833076 │ │ │ *
INVITE (SDP) * │ │Contact: <sip:68983619@3.236.72.101:5060>
+0.004863 │ │ │
*──────────────────────────>* │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
01:22:15.837939 │ │ │
100 Trying │ │CSeq: 102 INVITE
+0.799120 │ │ │
<────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0
01:22:16.637059 │ │ │ 183
Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT
+0.000179 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Supported: replaces, timer
+1.490537 │ <────────────────────────── │ │
│ │P-Preferred-Identity: <sip:68983600@10.0.76.9>
01:22:18.127775 │ │ │
180 Ringing │ │Content-Type: application/sdp
+0.000189 │ │ │
<────────────────────────── │ │Content-Length: 279
01:22:18.127964 │ 180 Ringing │ │
│ │
+0.349351 │ <────────────────────────── │ │
│ │v=0
01:22:18.477315 │ │ │
180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP
Proxy's private IP)*
+0.000206 │ │ │
<<<──────────────────────── │ │s=Asterisk PBX 17.7.0
01:22:18.477521 │ 180 Ringing │ │
│ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)*
+19.181387 │ <<<──────────────────────── │
│ │ │t=0 0
01:22:37.658908 │ │ │
200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101
+0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:8 PCMA/8000
01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=rtpmap:0 PCMU/8000
+0.241852 │ <────────────────────────── │ │
│ │a=rtpmap:101 telephone-event/8000
01:22:37.974479 │ ACK │ │
│ │a=fmtp:101 0-16
+0.000282 │ ──────────────────────────> │ │
│ │a=maxptime:150
01:22:37.974761 │ │ │
ACK │ │a=sendrecv
+4.095171 │ │ │
──────────────────────────> │ │a=nortpproxy:yes
01:22:42.069932 │ │ │
BYE │ │
+0.000361 │ │ │
<────────────────────────── │ │
01:22:42.070293 │ BYE │ │
│ │
+0.244125 │ <────────────────────────── │ │
│ │
01:22:42.314418 │ 200 OK │ │
│ │
+0.000275 │ ──────────────────────────> │ │
│ │
01:22:42.314693 │ │ │
200 OK │ │
│ │ │
──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060
10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Record-Route:
<sip:192.168.0.192;lr;ftag=as69eb1cce>
01:22:15.782149 │ INVITE (SDP) │ │
│ │Call-ID: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
+0.050579 │ ──────────────────────────> │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
01:22:15.832728 │ 100 trying -- your call is │ │
│ │To:
<sip:09413745250@192.168.0.192:5060>;tag=aa2c806-t7ln3f58c3ea1
+0.000348 │ <────────────────────────── │ │
│ │CSeq: 102 INVITE
01:22:15.833076 │ │ │
INVITE (SDP) │ │Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
+0.004863 │ │ │
──────────────────────────> │ │Contact:
<sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
01:22:15.837939 │ │ │
100 Trying │ │User-Agent: ZTE Softswitch/1.0.0
+0.799120 │ │ │
<────────────────────────── │ │Require: timer
01:22:16.637059 │ │ │ 183
Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac
+0.000179 │ │ │
<────────────────────────── │ │Content-Length: 208
01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
+1.490537 │ <────────────────────────── │ │
│ │
01:22:18.127775 │ │ │
180 Ringing │ │v=0
+0.000189 │ │ │
<────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP
Proxy's public IP)*
01:22:18.127964 │ 180 Ringing │ │
│ │s=SBC call
+0.349351 │ <────────────────────────── │ │
│ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)*
01:22:18.477315 │ │ │
180 Ringing │ │t=0 0
+0.000206 │ │ │
<<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101
01:22:18.477521 │ 180 Ringing │ │
│ │a=rtpmap:101 telephone-event/8000
+19.181387 │ <<<──────────────────────── │
│ │ │a=fmtp:101 0-15
01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1
+0.073719 │ │ │
<────────────────────────── │ │a=nortpproxy:yes
01:22:37.732627 │ * 200 OK (SDP)* │ │
│ │
+0.241852 │ *<──────────────────────────* │ │
│ │
01:22:37.974479 │ ACK │ │
│ │
+0.000282 │ ──────────────────────────> │ │
│ │
01:22:37.974761 │ │ │
ACK │ │
+4.095171 │ │ │
──────────────────────────> │ │
01:22:42.069932 │ │ │
BYE │ │
+0.000361 │ │ │
<────────────────────────── │ │
01:22:42.070293 │ BYE │ │
│ │
+0.244125 │ <────────────────────────── │ │
│ │
01:22:42.314418 │ 200 OK │ │
│ │
+0.000275 │ ──────────────────────────> │ │
│ │
01:22:42.314693 │ │ │
200 OK │ │
│ │ │
──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command:
*/usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -u
asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230
<http://192.168.0.192/10.0.87.230>*
Apart from this, in the Asterisk console I can see that the RTP packets
are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for
incoming calls
Anything am I missing here?
Thanks.
Regards
Kashish
>
8:13 p.m.
Kashish,
Best you can do is take a trace on BOTH ends and share it with us.
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
On Thu, May 27, 2021 at 2:26 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Haven't been able to sort this out yet. Anything
am I missing here?
Thanks.
Regards
Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
Hi Daniel,
Sorry it took some time for me to make these changes.
I have made all the changes as suggested by you however it still doesn't
seem to work. No audio in the outbound call however incoming call works
fine.
Here are the SIP traces after making the changes:
*INVITE: Asterisk to Kamailio:*
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060
10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;rport
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Max-Forwards: 70
01:22:15.782149 │ *INVITE (SDP) * │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
+0.050579 │ *──────────────────────────>* │ │
│ │To: <sip:09413745250@192.168.0.192:5060>
01:22:15.832728 │ 100 trying -- your call is │ │
│ │Contact: <sip:68983619@3.236.72.101:5060>
+0.000348 │ <────────────────────────── │ │
│ │Call-ID: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
01:22:15.833076 │ │ │
INVITE (SDP) │ │CSeq: 102 INVITE
+0.004863 │ │ │
──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0
01:22:15.837939 │ │ │
100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT
+0.799120 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637059 │ │ │ 183
Session Progress (SDP) │ │Supported: replaces, timer
+0.000179 │ │ │
<────────────────────────── │ │P-Preferred-Identity:
<sip:68983600@10.0.76.9>
01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
+1.490537 │ <────────────────────────── │ │
│ │Content-Length: 263
01:22:18.127775 │ │ │
180 Ringing │ │
+0.000189 │ │ │
<────────────────────────── │ │v=0
01:22:18.127964 │ 180 Ringing │ │
│ │o=root 1560151942 1560151942 IN IP4 3.236.72.101
*(Asterisk's Public IP)*
+0.349351 │ <────────────────────────── │ │
│ │s=Asterisk PBX 17.7.0
01:22:18.477315 │ │ │
180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)*
+0.000206 │ │ │
<<<──────────────────────── │ │t=0 0
01:22:18.477521 │ 180 Ringing │ │
│ │m=audio 14046 RTP/AVP 8 0 101
+19.181387 │ <<<──────────────────────── │
│ │ │a=rtpmap:8 PCMA/8000
01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:0 PCMU/8000
+0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:101 telephone-event/8000
01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=fmtp:101 0-16
+0.241852 │ <────────────────────────── │ │
│ │a=maxptime:150
01:22:37.974479 │ ACK │ │
│ │a=sendrecv
+0.000282 │ ──────────────────────────> │ │
│ │
01:22:37.974761 │ │ │
ACK │ │
+4.095171 │ │ │
──────────────────────────> │ │
01:22:42.069932 │ │ │
BYE │ │
+0.000361 │ │ │
<────────────────────────── │ │
01:22:42.070293 │ BYE │ │
│ │
+0.244125 │ <────────────────────────── │ │
│ │
01:22:42.314418 │ 200 OK │ │
│ │
+0.000275 │ ──────────────────────────> │ │
│ │
01:22:42.314693 │ │ │
200 OK │ │
│ │ │
──────────────────────────> │ │
*INVITE: Kamailio to Telco:*
│INVITE sip:09413745250@10.0.76.9 SIP/2.0
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060
10.0.76.9:5060 │Record-Route:
<sip:192.168.0.192;lr=on;ftag=as69eb1cce>
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0
01:22:15.782149 │ INVITE (SDP) │ │
│ │Via: SIP/2.0/UDP
3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060
+0.050579 │ ──────────────────────────> │ │
│ │Max-Forwards: 69
01:22:15.832728 │ 100 trying -- your call is │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
+0.000348 │ <────────────────────────── │ │
│ │To: <sip:09413745250@192.168.0.192:5060>
01:22:15.833076 │ │ │ *
INVITE (SDP) * │ │Contact: <sip:68983619@3.236.72.101:5060>
+0.004863 │ │ │
*──────────────────────────>* │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
01:22:15.837939 │ │ │
100 Trying │ │CSeq: 102 INVITE
+0.799120 │ │ │
<────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0
01:22:16.637059 │ │ │ 183
Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT
+0.000179 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Supported: replaces, timer
+1.490537 │ <────────────────────────── │ │
│ │P-Preferred-Identity: <sip:68983600@10.0.76.9>
01:22:18.127775 │ │ │
180 Ringing │ │Content-Type: application/sdp
+0.000189 │ │ │
<────────────────────────── │ │Content-Length: 279
01:22:18.127964 │ 180 Ringing │ │
│ │
+0.349351 │ <────────────────────────── │ │
│ │v=0
01:22:18.477315 │ │ │
180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP
Proxy's private IP)*
+0.000206 │ │ │
<<<──────────────────────── │ │s=Asterisk PBX 17.7.0
01:22:18.477521 │ 180 Ringing │ │
│ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)*
+19.181387 │ <<<──────────────────────── │
│ │ │t=0 0
01:22:37.658908 │ │ │
200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101
+0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:8 PCMA/8000
01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=rtpmap:0 PCMU/8000
+0.241852 │ <────────────────────────── │ │
│ │a=rtpmap:101 telephone-event/8000
01:22:37.974479 │ ACK │ │
│ │a=fmtp:101 0-16
+0.000282 │ ──────────────────────────> │ │
│ │a=maxptime:150
01:22:37.974761 │ │ │
ACK │ │a=sendrecv
+4.095171 │ │ │
──────────────────────────> │ │a=nortpproxy:yes
01:22:42.069932 │ │ │
BYE │ │
+0.000361 │ │ │
<────────────────────────── │ │
01:22:42.070293 │ BYE │ │
│ │
+0.244125 │ <────────────────────────── │ │
│ │
01:22:42.314418 │ 200 OK │ │
│ │
+0.000275 │ ──────────────────────────> │ │
│ │
01:22:42.314693 │ │ │
200 OK │ │
│ │ │
──────────────────────────> │ │
*On 200: Kamailio to Asterisk:*
│SIP/2.0 200 OK
3.236.72.101:5060 192.168.0.192:5060 10.0.87.230:5060
10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060
──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Record-Route:
<sip:192.168.0.192;lr;ftag=as69eb1cce>
01:22:15.782149 │ INVITE (SDP) │ │
│ │Call-ID: 1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
+0.050579 │ ──────────────────────────> │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
01:22:15.832728 │ 100 trying -- your call is │ │
│ │To:
<sip:09413745250@192.168.0.192:5060>;tag=aa2c806-t7ln3f58c3ea1
+0.000348 │ <────────────────────────── │ │
│ │CSeq: 102 INVITE
01:22:15.833076 │ │ │
INVITE (SDP) │ │Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
+0.004863 │ │ │
──────────────────────────> │ │Contact:
<sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
01:22:15.837939 │ │ │
100 Trying │ │User-Agent: ZTE Softswitch/1.0.0
+0.799120 │ │ │
<────────────────────────── │ │Require: timer
01:22:16.637059 │ │ │ 183
Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac
+0.000179 │ │ │
<────────────────────────── │ │Content-Length: 208
01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
+1.490537 │ <────────────────────────── │ │
│ │
01:22:18.127775 │ │ │
180 Ringing │ │v=0
+0.000189 │ │ │
<────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP
Proxy's public IP)*
01:22:18.127964 │ 180 Ringing │ │
│ │s=SBC call
+0.349351 │ <────────────────────────── │ │
│ │c=IN IP4 192.168.0.192 *(RTP Proxy's public IP)*
01:22:18.477315 │ │ │
180 Ringing │ │t=0 0
+0.000206 │ │ │
<<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101
01:22:18.477521 │ 180 Ringing │ │
│ │a=rtpmap:101 telephone-event/8000
+19.181387 │ <<<──────────────────────── │
│ │ │a=fmtp:101 0-15
01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1
+0.073719 │ │ │
<────────────────────────── │ │a=nortpproxy:yes
01:22:37.732627 │ * 200 OK (SDP)* │ │
│ │
+0.241852 │ *<──────────────────────────* │ │
│ │
01:22:37.974479 │ ACK │ │
│ │
+0.000282 │ ──────────────────────────> │ │
│ │
01:22:37.974761 │ │ │
ACK │ │
+4.095171 │ │ │
──────────────────────────> │ │
01:22:42.069932 │ │ │
BYE │ │
+0.000361 │ │ │
<────────────────────────── │ │
01:22:42.070293 │ BYE │ │
│ │
+0.244125 │ <────────────────────────── │ │
│ │
01:22:42.314418 │ 200 OK │ │
│ │
+0.000275 │ ──────────────────────────> │ │
│ │
01:22:42.314693 │ │ │
200 OK │ │
│ │ │
──────────────────────────> │ │
On the cloud Asterisk, all the relevant public IPs are already allowed.
Have run the rtpproxy on the bridge mode with the following command:
*/usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -u
asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230
<http://192.168.0.192/10.0.87.230>*
Apart from this, in the Asterisk console I can see that the RTP packets
are being sent to Kamailio
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len 000160)
Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len 000160)
However, there isn't any log for receiving the RTP packets unlike for
incoming calls
Anything am I missing here?
Thanks.
Regards
Kashish
> __________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
the sender!
Edit mailing list options or unsubscribe:
* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
11:05 p.m.
Yes, have shared the complete traces in this thread itself.
Thanks.
Regards
Kashish
On Thu, May 27, 2021 at 6:44 PM Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Haven't been able to sort this out yet. Anything
am I missing here?
Thanks.
Regards
Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
> Hi Daniel,
>
> Sorry it took some time for me to make these changes.
>
> I have made all the changes as suggested by you however it still doesn't
> seem to work. No audio in the outbound call however incoming call works
> fine.
>
> Here are the SIP traces after making the changes:
>
> *INVITE: Asterisk to Kamailio:*
>
>
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0
> 3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;rport
> ──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Max-Forwards: 70
> 01:22:15.782149 │ *INVITE (SDP) * │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
> +0.050579 │ *──────────────────────────>* │
│ │ │To: <sip:09413745250@192.168.0.192:5060>
> 01:22:15.832728 │ 100 trying -- your call is │ │
│ │Contact: <sip:68983619@3.236.72.101:5060>
> +0.000348 │ <────────────────────────── │ │
│ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
> 01:22:15.833076 │ │ │
INVITE (SDP) │ │CSeq: 102 INVITE
> +0.004863 │ │ │
──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0
> 01:22:15.837939 │ │ │
100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT
> +0.799120 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> 01:22:16.637059 │ │ │
183 Session Progress (SDP) │ │Supported: replaces, timer
> +0.000179 │ │ │
<────────────────────────── │ │P-Preferred-Identity:
<sip:68983600@10.0.76.9>
> 01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
> +1.490537 │ <────────────────────────── │ │
│ │Content-Length: 263
> 01:22:18.127775 │ │ │
180 Ringing │ │
> +0.000189 │ │ │
<────────────────────────── │ │v=0
> 01:22:18.127964 │ 180 Ringing │ │
│ │o=root 1560151942 1560151942 IN IP4 3.236.72.101
*(Asterisk's Public IP)*
> +0.349351 │ <────────────────────────── │ │
│ │s=Asterisk PBX 17.7.0
> 01:22:18.477315 │ │ │
180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)*
> +0.000206 │ │ │
<<<──────────────────────── │ │t=0 0
> 01:22:18.477521 │ 180 Ringing │ │
│ │m=audio 14046 RTP/AVP 8 0 101
> +19.181387 │ <<<──────────────────────── │
│ │ │a=rtpmap:8 PCMA/8000
> 01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:0 PCMU/8000
> +0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:101 telephone-event/8000
> 01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=fmtp:101 0-16
> +0.241852 │ <────────────────────────── │ │
│ │a=maxptime:150
> 01:22:37.974479 │ ACK │ │
│ │a=sendrecv
> +0.000282 │ ──────────────────────────> │ │
│ │
> 01:22:37.974761 │ │ │
ACK │ │
> +4.095171 │ │ │
──────────────────────────> │ │
> 01:22:42.069932 │ │ │
BYE │ │
> +0.000361 │ │ │
<────────────────────────── │ │
> 01:22:42.070293 │ BYE │ │
│ │
> +0.244125 │ <────────────────────────── │ │
│ │
> 01:22:42.314418 │ 200 OK │ │
│ │
> +0.000275 │ ──────────────────────────> │ │
│ │
> 01:22:42.314693 │ │ │
200 OK │ │
> │ │ │
──────────────────────────> │ │
>
>
>
> *INVITE: Kamailio to Telco:*
>
>
│INVITE sip:09413745250@10.0.76.9 SIP/2.0
> 3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Record-Route:
<sip:192.168.0.192;lr=on;ftag=as69eb1cce>
> ──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0
> 01:22:15.782149 │ INVITE (SDP) │ │
│ │Via: SIP/2.0/UDP
3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060
> +0.050579 │ ──────────────────────────> │ │
│ │Max-Forwards: 69
> 01:22:15.832728 │ 100 trying -- your call is │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
> +0.000348 │ <────────────────────────── │ │
│ │To: <sip:09413745250@192.168.0.192:5060>
> 01:22:15.833076 │ │ │ *
INVITE (SDP) * │ │Contact: <sip:68983619@3.236.72.101:5060>
> +0.004863 │ │ │
*──────────────────────────>* │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
> 01:22:15.837939 │ │ │
100 Trying │ │CSeq: 102 INVITE
> +0.799120 │ │ │
<────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0
> 01:22:16.637059 │ │ │
183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT
> +0.000179 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> 01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Supported: replaces, timer
> +1.490537 │ <────────────────────────── │ │
│ │P-Preferred-Identity:
<sip:68983600@10.0.76.9>
> 01:22:18.127775 │ │ │
180 Ringing │ │Content-Type: application/sdp
> +0.000189 │ │ │
<────────────────────────── │ │Content-Length: 279
> 01:22:18.127964 │ 180 Ringing │ │
│ │
> +0.349351 │ <────────────────────────── │ │
│ │v=0
> 01:22:18.477315 │ │ │
180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP
Proxy's private IP)*
> +0.000206 │ │ │
<<<──────────────────────── │ │s=Asterisk PBX 17.7.0
> 01:22:18.477521 │ 180 Ringing │ │
│ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)*
> +19.181387 │ <<<──────────────────────── │
│ │ │t=0 0
> 01:22:37.658908 │ │ │
200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101
> +0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:8 PCMA/8000
> 01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=rtpmap:0 PCMU/8000
> +0.241852 │ <────────────────────────── │ │
│ │a=rtpmap:101 telephone-event/8000
> 01:22:37.974479 │ ACK │ │
│ │a=fmtp:101 0-16
> +0.000282 │ ──────────────────────────> │ │
│ │a=maxptime:150
> 01:22:37.974761 │ │ │
ACK │ │a=sendrecv
> +4.095171 │ │ │
──────────────────────────> │ │a=nortpproxy:yes
> 01:22:42.069932 │ │ │
BYE │ │
> +0.000361 │ │ │
<────────────────────────── │ │
> 01:22:42.070293 │ BYE │ │
│ │
> +0.244125 │ <────────────────────────── │ │
│ │
> 01:22:42.314418 │ 200 OK │ │
│ │
> +0.000275 │ ──────────────────────────> │ │
│ │
> 01:22:42.314693 │ │ │
200 OK │ │
> │ │ │
──────────────────────────> │ │
>
>
>
> *On 200: Kamailio to Asterisk:*
>
>
│SIP/2.0 200 OK
> 3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060
> ──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Record-Route:
<sip:192.168.0.192;lr;ftag=as69eb1cce>
> 01:22:15.782149 │ INVITE (SDP) │ │
│ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
> +0.050579 │ ──────────────────────────> │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
> 01:22:15.832728 │ 100 trying -- your call is │ │
│ │To:
<sip:09413745250@192.168.0.192:5060>;tag=aa2c806-t7ln3f58c3ea1
> +0.000348 │ <────────────────────────── │ │
│ │CSeq: 102 INVITE
> 01:22:15.833076 │ │ │
INVITE (SDP) │ │Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> +0.004863 │ │ │
──────────────────────────> │ │Contact:
<sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> 01:22:15.837939 │ │ │
100 Trying │ │User-Agent: ZTE Softswitch/1.0.0
> +0.799120 │ │ │
<────────────────────────── │ │Require: timer
> 01:22:16.637059 │ │ │
183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac
> +0.000179 │ │ │
<────────────────────────── │ │Content-Length: 208
> 01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
> +1.490537 │ <────────────────────────── │ │
│ │
> 01:22:18.127775 │ │ │
180 Ringing │ │v=0
> +0.000189 │ │ │
<────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP
Proxy's public IP)*
> 01:22:18.127964 │ 180 Ringing │ │
│ │s=SBC call
> +0.349351 │ <────────────────────────── │ │
│ │c=IN IP4 192.168.0.192 *(RTP Proxy's public
IP)*
> 01:22:18.477315 │ │ │
180 Ringing │ │t=0 0
> +0.000206 │ │ │
<<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101
> 01:22:18.477521 │ 180 Ringing │ │
│ │a=rtpmap:101 telephone-event/8000
> +19.181387 │ <<<──────────────────────── │
│ │ │a=fmtp:101 0-15
> 01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1
> +0.073719 │ │ │
<────────────────────────── │ │a=nortpproxy:yes
> 01:22:37.732627 │ * 200 OK (SDP)* │ │
│ │
> +0.241852 │ *<──────────────────────────* │
│ │ │
> 01:22:37.974479 │ ACK │ │
│ │
> +0.000282 │ ──────────────────────────> │ │
│ │
> 01:22:37.974761 │ │ │
ACK │ │
> +4.095171 │ │ │
──────────────────────────> │ │
> 01:22:42.069932 │ │ │
BYE │ │
> +0.000361 │ │ │
<────────────────────────── │ │
> 01:22:42.070293 │ BYE │ │
│ │
> +0.244125 │ <────────────────────────── │ │
│ │
> 01:22:42.314418 │ 200 OK │ │
│ │
> +0.000275 │ ──────────────────────────> │ │
│ │
> 01:22:42.314693 │ │ │
200 OK │ │
> │ │ │
──────────────────────────> │ │
>
>
>
> On the cloud Asterisk, all the relevant public IPs are already allowed.
>
> Have run the rtpproxy on the bridge mode with the following command:
> */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 <http://127.0.0.1:7722> -u
> asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230
> <http://192.168.0.192/10.0.87.230>*
>
> Apart from this, in the Asterisk console I can see that the RTP packets
> are being sent to Kamailio
>
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len
000160)
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len
000160)
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len
000160)
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len
000160)
>
>
> However, there isn't any log for receiving the RTP packets unlike for
> incoming calls
>
> Anything am I missing here?
>
> Thanks.
> Regards
> Kashish
>
>>
28 May
28 May
11:29 a.m.
It’s very difficult to read, don’t paste an sngrep screenshot. If you can,
please save the call to a file, then share that.
:)
On Thu, 27 May 2021 at 21:09, Kashish Raheja <kashishraheja1809(a)gmail.com>
wrote:
Yes, have shared the complete traces in this thread
itself.
Thanks.
Regards
Kashish
On Thu, May 27, 2021 at 6:44 PM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Haven't been able to sort this out yet.
Anything am I missing here?
Thanks.
Regards
Kashish
On Fri, May 21, 2021 at 1:44 AM Kashish Raheja <
kashishraheja1809(a)gmail.com> wrote:
> Hi Daniel,
>
> Sorry it took some time for me to make these changes.
>
> I have made all the changes as suggested by you however it still doesn't
> seem to work. No audio in the outbound call however incoming call works
> fine.
>
> Here are the SIP traces after making the changes:
>
> *INVITE: Asterisk to Kamailio:*
>
>
│INVITE sip:09413745250@192.168.0.192:5060 SIP/2.0
> 3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;rport
> ──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Max-Forwards: 70
> 01:22:15.782149 │ *INVITE (SDP) * │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
> +0.050579 │ *──────────────────────────>* │
│ │ │To: <sip:09413745250@192.168.0.192:5060>
> 01:22:15.832728 │ 100 trying -- your call is │ │
│ │Contact: <sip:68983619@3.236.72.101:5060>
> +0.000348 │ <────────────────────────── │ │
│ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
> 01:22:15.833076 │ │ │
INVITE (SDP) │ │CSeq: 102 INVITE
> +0.004863 │ │ │
──────────────────────────> │ │User-Agent: Asterisk PBX 17.7.0
> 01:22:15.837939 │ │ │
100 Trying │ │Date: Thu, 20 May 2021 19:52:15 GMT
> +0.799120 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> 01:22:16.637059 │ │ │
183 Session Progress (SDP) │ │Supported: replaces, timer
> +0.000179 │ │ │
<────────────────────────── │ │P-Preferred-Identity:
<sip:68983600@10.0.76.9>
> 01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
> +1.490537 │ <────────────────────────── │ │
│ │Content-Length: 263
> 01:22:18.127775 │ │ │
180 Ringing │ │
> +0.000189 │ │ │
<────────────────────────── │ │v=0
> 01:22:18.127964 │ 180 Ringing │ │
│ │o=root 1560151942 1560151942 IN IP4 3.236.72.101
*(Asterisk's Public IP)*
> +0.349351 │ <────────────────────────── │ │
│ │s=Asterisk PBX 17.7.0
> 01:22:18.477315 │ │ │
180 Ringing │ │c=IN IP4 3.236.72.101 *(Asterisk's Public IP)*
> +0.000206 │ │ │
<<<──────────────────────── │ │t=0 0
> 01:22:18.477521 │ 180 Ringing │ │
│ │m=audio 14046 RTP/AVP 8 0 101
> +19.181387 │ <<<──────────────────────── │
│ │ │a=rtpmap:8 PCMA/8000
> 01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:0 PCMU/8000
> +0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:101 telephone-event/8000
> 01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=fmtp:101 0-16
> +0.241852 │ <────────────────────────── │ │
│ │a=maxptime:150
> 01:22:37.974479 │ ACK │ │
│ │a=sendrecv
> +0.000282 │ ──────────────────────────> │ │
│ │
> 01:22:37.974761 │ │ │
ACK │ │
> +4.095171 │ │ │
──────────────────────────> │ │
> 01:22:42.069932 │ │ │
BYE │ │
> +0.000361 │ │ │
<────────────────────────── │ │
> 01:22:42.070293 │ BYE │ │
│ │
> +0.244125 │ <────────────────────────── │ │
│ │
> 01:22:42.314418 │ 200 OK │ │
│ │
> +0.000275 │ ──────────────────────────> │ │
│ │
> 01:22:42.314693 │ │ │
200 OK │ │
> │ │ │
──────────────────────────> │ │
>
>
>
> *INVITE: Kamailio to Telco:*
>
>
│INVITE sip:09413745250@10.0.76.9 SIP/2.0
> 3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Record-Route:
<sip:192.168.0.192;lr=on;ftag=as69eb1cce>
> ──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Via: SIP/2.0/UDP
192.168.0.192;branch=z9hG4bK9a7.1dc719cb9791c895364af0a28d633d02.0
> 01:22:15.782149 │ INVITE (SDP) │ │
│ │Via: SIP/2.0/UDP
3.236.72.101:5060;received=3.236.72.101;branch=z9hG4bK62f0d772;rport=5060
> +0.050579 │ ──────────────────────────> │ │
│ │Max-Forwards: 69
> 01:22:15.832728 │ 100 trying -- your call is │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
> +0.000348 │ <────────────────────────── │ │
│ │To: <sip:09413745250@192.168.0.192:5060>
> 01:22:15.833076 │ │ │ *
INVITE (SDP) * │ │Contact: <sip:68983619@3.236.72.101:5060>
> +0.004863 │ │ │
*──────────────────────────>* │ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
> 01:22:15.837939 │ │ │
100 Trying │ │CSeq: 102 INVITE
> +0.799120 │ │ │
<────────────────────────── │ │User-Agent: Asterisk PBX 17.7.0
> 01:22:16.637059 │ │ │
183 Session Progress (SDP) │ │Date: Thu, 20 May 2021 19:52:15 GMT
> +0.000179 │ │ │
<────────────────────────── │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> 01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Supported: replaces, timer
> +1.490537 │ <────────────────────────── │ │
│ │P-Preferred-Identity:
<sip:68983600@10.0.76.9>
> 01:22:18.127775 │ │ │
180 Ringing │ │Content-Type: application/sdp
> +0.000189 │ │ │
<────────────────────────── │ │Content-Length: 279
> 01:22:18.127964 │ 180 Ringing │ │
│ │
> +0.349351 │ <────────────────────────── │ │
│ │v=0
> 01:22:18.477315 │ │ │
180 Ringing │ │o=root 1560151942 1560151942 IN IP4 10.0.87.230 *(RTP
Proxy's private IP)*
> +0.000206 │ │ │
<<<──────────────────────── │ │s=Asterisk PBX 17.7.0
> 01:22:18.477521 │ 180 Ringing │ │
│ │c=IN IP4 10.0.87.230 *(RTP Proxy's private IP)*
> +19.181387 │ <<<──────────────────────── │
│ │ │t=0 0
> 01:22:37.658908 │ │ │
200 OK (SDP) │ │m=audio 37322 RTP/AVP 8 0 101
> +0.073719 │ │ │
<────────────────────────── │ │a=rtpmap:8 PCMA/8000
> 01:22:37.732627 │ 200 OK (SDP) │ │
│ │a=rtpmap:0 PCMU/8000
> +0.241852 │ <────────────────────────── │ │
│ │a=rtpmap:101 telephone-event/8000
> 01:22:37.974479 │ ACK │ │
│ │a=fmtp:101 0-16
> +0.000282 │ ──────────────────────────> │ │
│ │a=maxptime:150
> 01:22:37.974761 │ │ │
ACK │ │a=sendrecv
> +4.095171 │ │ │
──────────────────────────> │ │a=nortpproxy:yes
> 01:22:42.069932 │ │ │
BYE │ │
> +0.000361 │ │ │
<────────────────────────── │ │
> 01:22:42.070293 │ BYE │ │
│ │
> +0.244125 │ <────────────────────────── │ │
│ │
> 01:22:42.314418 │ 200 OK │ │
│ │
> +0.000275 │ ──────────────────────────> │ │
│ │
> 01:22:42.314693 │ │ │
200 OK │ │
> │ │ │
──────────────────────────> │ │
>
>
>
> *On 200: Kamailio to Asterisk:*
>
>
│SIP/2.0 200 OK
> 3.236.72.101:5060 192.168.0.192:5060
10.0.87.230:5060 10.0.76.9:5060 │Via: SIP/2.0/UDP
3.236.72.101:5060;branch=z9hG4bK62f0d772;received=3.236.72.101;rport=5060
> ──────────┬───────── ──────────┬─────────
──────────┬───────── ──────────┬─────────│Record-Route:
<sip:192.168.0.192;lr;ftag=as69eb1cce>
> 01:22:15.782149 │ INVITE (SDP) │ │
│ │Call-ID:
1191aedf331ec3e35955bf376a20999d(a)14.98.22.110
> +0.050579 │ ──────────────────────────> │ │
│ │From:
<sip:68983619@192.168.0.192:5060>;tag=as69eb1cce
> 01:22:15.832728 │ 100 trying -- your call is │ │
│ │To:
<sip:09413745250@192.168.0.192:5060>;tag=aa2c806-t7ln3f58c3ea1
> +0.000348 │ <────────────────────────── │ │
│ │CSeq: 102 INVITE
> 01:22:15.833076 │ │ │
INVITE (SDP) │ │Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
> +0.004863 │ │ │
──────────────────────────> │ │Contact:
<sip:09413745250@10.0.76.9:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
> 01:22:15.837939 │ │ │
100 Trying │ │User-Agent: ZTE Softswitch/1.0.0
> +0.799120 │ │ │
<────────────────────────── │ │Require: timer
> 01:22:16.637059 │ │ │
183 Session Progress (SDP) │ │Session-Expires: 7200;refresher=uac
> +0.000179 │ │ │
<────────────────────────── │ │Content-Length: 208
> 01:22:16.637238 │ 183 Session Progress (SDP) │ │
│ │Content-Type: application/sdp
> +1.490537 │ <────────────────────────── │ │
│ │
> 01:22:18.127775 │ │ │
180 Ringing │ │v=0
> +0.000189 │ │ │
<────────────────────────── │ │o=- 1026 13186 IN IP4 192.168.0.192 *(RTP
Proxy's public IP)*
> 01:22:18.127964 │ 180 Ringing │ │
│ │s=SBC call
> +0.349351 │ <────────────────────────── │ │
│ │c=IN IP4 192.168.0.192 *(RTP Proxy's public
IP)*
> 01:22:18.477315 │ │ │
180 Ringing │ │t=0 0
> +0.000206 │ │ │
<<<──────────────────────── │ │m=audio 48462 RTP/AVP 8 101
> 01:22:18.477521 │ 180 Ringing │ │
│ │a=rtpmap:101 telephone-event/8000
> +19.181387 │ <<<──────────────────────── │
│ │ │a=fmtp:101 0-15
> 01:22:37.658908 │ │ │
200 OK (SDP) │ │a=rtpmap:8 PCMA/8000/1
> +0.073719 │ │ │
<────────────────────────── │ │a=nortpproxy:yes
> 01:22:37.732627 │ * 200 OK (SDP)* │ │
│ │
> +0.241852 │ *<──────────────────────────* │
│ │ │
> 01:22:37.974479 │ ACK │ │
│ │
> +0.000282 │ ──────────────────────────> │ │
│ │
> 01:22:37.974761 │ │ │
ACK │ │
> +4.095171 │ │ │
──────────────────────────> │ │
> 01:22:42.069932 │ │ │
BYE │ │
> +0.000361 │ │ │
<────────────────────────── │ │
> 01:22:42.070293 │ BYE │ │
│ │
> +0.244125 │ <────────────────────────── │ │
│ │
> 01:22:42.314418 │ 200 OK │ │
│ │
> +0.000275 │ ──────────────────────────> │ │
│ │
> 01:22:42.314693 │ │ │
200 OK │ │
> │ │ │
──────────────────────────> │ │
>
>
>
> On the cloud Asterisk, all the relevant public IPs are already allowed.
>
> Have run the rtpproxy on the bridge mode with the following command:
> */usr/local/bin/rtpproxy -s udp:127.0.0.1:7722 <http://127.0.0.1:7722>
> -u asterisk -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.192/10.0.87.230
> <http://192.168.0.192/10.0.87.230>*
>
> Apart from this, in the Asterisk console I can see that the RTP packets
> are being sent to Kamailio
>
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022310, ts 029280, len
000160)
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022311, ts 029440, len
000160)
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022312, ts 029600, len
000160)
> Sent RTP packet to 14.98.22.110:11648 (type 08, seq 022313, ts 029760, len
000160)
>
>
> However, there isn't any log for receiving the RTP packets unlike for
> incoming calls
>
> Anything am I missing here?
>
> Thanks.
> Regards
> Kashish
>
>> __________________________________________________________
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1274
days inactive
1292
days old
15 comments
4 participants
participants (4)
-
Daniel Donoghue -
David Villasmil -
Kashish Raheja -
Sergio Charrua