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Issue in voice transfer - media server

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harikumarms
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Joined: 2008-09-08
Points: 0

Hi All,

I have an issue in voice transferring between two client in differnt LAN.

When I call from UAC1 to UAC2 where both are in the same LAN, then call is establishing and voice is transferring.

UAC1 -----> NAT -----> Internet ----> Mobicent server(Static IP)
UAC2 -------->'

When I call from UAC1(in Static IP) and UAC2(behind NAT), call is establishing and voice is transferring and vice versa also working fine.

UAC1(Static IP) --->Internet --->Mobicent server(Static IP)<----Internet <---NAT<---UAC2

When I call from UAC1(behind NAT) and UAC2(behind different NAT), call is established but voice is not transferring.

UAC1 --->NAT--->Internet --->Mobicent server(Static IP)<----Internet <---NAT<---UAC2

How can I fix the issue, I configure the server.xml for STUN also.

I have a doubt whether voice packet is transferring through media proxy server or it is transferring peer to peer.

If it is peer to peer, then what configuration change I have to make in order to transfer the voice packet through media serrver, I have installated the media server.

Is there is any way to identify whether voice packet is transferring through media server, because for voice packet transferring I couldnt get any log.

Please suggest me to fix out the above issue.

Thanks
M.S.HariKumar

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harikumarms
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Joined: 2008-09-08
Points: 0

Hi All,

I have verified the packet in UAC and Server side for the above scenario.
[u][b]
In the first scenario:[/b][/u]

UAC1(Public IP - 123.246.244.50) -------> Mobicent Media Server(Public IP - 70.32.121.36).

[b]a) UAC sends packet to server.[/b]

[b]SIP Signal Packet Information:[/b]
Source IP: 123.246.244.50, Port: 54204
Destination IP: 70.32.121.36, Port: 5090
[b]
RTP Packet Information:[/b]
Source IP: 123.246.244.50, Port: 38838
Destination IP: 70.32.121.36, Port: 60000

[b]b) Server sends packet to UAC[/b]

[b]SIP Signal Packet Information:[/b]
Source IP: 70.32.121.36, Port: 5090
Destination IP: 123.246.244.50, Port: 54204

[b]RTP Packet Information:[/b]
Source IP: 70.32.121.36, Port: 60000
Destination IP: 123.246.244.50, Port: 38838

In the above scenario both Sip and RTP packet are receiving in both end.

[u][b]In the second scenario:[/b][/u]

UAC1(Private IP - 192.168.1.100) ---->NAT(Public IP - 123.246.244.50)-------> Mobicent Media Server(Public IP - 70.32.121.36)

[b]a) UAC sends packet to server.[/b]

[b]SIP Signal Packet Information:[/b]
Source IP: 123.246.244.50, Port: 54204
Destination IP: 70.32.121.36, Port: 5090

[b]RTP Packet Information:[/b]
Source IP: 123.246.244.50, Port: 38838
Destination IP: 70.32.121.36, Port: 60000

[b]b) Server sends packet to UAC[/b]
SIP Signal Packet Information:
Source IP: 70.32.121.36, Port: 5090
Destination IP: 123.246.244.50, Port: 54204
[b]
RTP Packet Information:[/b]
Source IP: 70.32.121.36, Port: 60000
[b]Destination IP: 192.168.1.100, Port: 38838[/b]

In the above scenario there is no problem sending the packets from UAC to server, it reaching successfully.
Similarly from the server Sip packets is received in UAC.
But when server sending RTP packet, it sends to UAC private IP address because of thim is not reache UAC.

Anybody please mention what changes I have to make inorder to pass the RTP packet to the destination UAC(behind NAT).

Thanks
M.S.HariKumar

harikumarms
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Joined: 2008-09-08
Points: 0

Is there is any configuration I have to make in server.xml to transfer the voice packet through media server?

deruelle_jean
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Joined: 2003-06-24
Points: 0

To be know where the media stream goes through. You 'll need to share with us the SDP exchanged during the call setup and the ip addresses of each UA and server.
My guess is that the SDP doesn't go through the Media Server. To allow this you'll need to use Packet Relay I think and the Sip Servlets Application will need to act as a B2BUA and modify the SDP exchanged.

Jean

harikumarms
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Joined: 2008-09-08
Points: 0

Hi Jean,

Thanks for your reply.
Do we have a sample code for using Packet Relay in call controller?
If so please share the source code or the code snippet.

Thanks,
M.S.HariKumar

kulikoff
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Joined: 2005-11-30
Points: 0

The mms-demo example explains how to use Packet Relay endpoint. You need to create two RTP connections on two PR endpoints and join endpoints using link

Regards,
Oleg

harikumarms
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Joined: 2008-09-08
Points: 0

Hi Oleg,

Please give me the suggestion to sort out the below issue.

[u][b]In MMS-Demo Example:[/b][/u]

[b][u]First Scenario[/u][/b]
UAC1(Public IP - 123.246.244.50) -------> Mobicent Media Server(Public IP - 70.32.121.36).

In the above scenario, when I dial 1010, I can hear the announcement and in loop back process I can hear my voice.

[u][b]Scenod Scenario[/b][/u]

UAC1(Private IP - 192.168.1.100) ---->NAT(Public IP - i123.246.244.50)-------> Mobicent Media Server(Public IP - 70.32.121.36)

In the second scenario, when I dial 1010, I can hear only Ringing sound. I couldn't hear announcement voice and my voice also.

Using Mobicent Media Server RTP packet will transfer to the Client behind the NAT, mobicent can able to handle the media packet behind NAT.

If Mobicent couldn't handle the packet, then give a suggestion to fix out the issue.

Thanks
M.S.HariKumar