11 May
2021
11 May
'21
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
>>
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