[SR-Users] No Audio For Outbound Calls

[David Villasmil]

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 at gmail.com
phone: +34669448337


On Mon, May 10, 2021 at 8:42 PM Kashish Raheja <kashishraheja1809 at 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 at 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 at 14.98.22.110
>> From: <sip:68XXXXX at 10.0.X.X>;tag=as2b21d944
>> To: <sip:09413745250 at 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 at 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 at 14.98.22.110
>> From: <sip:68XXXXX at 10.0.X.X>;tag=as2b21d944
>> To: <sip:09413745250 at 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 at 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 at 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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