11 May
2021
11 May
'21
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
__________________________________________________________
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