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Questions on RTP

 
 
Steve
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      12-01-2008
Hi,

I configured my Asterisk server to send RTP wiht the port range : 30000
- 31000 ]
This way I can priotize the communication with the QoS

Now, last time I received a phone call, coming from my provider, which
I suppose got an Asterisk server as well and the COM was with a long
delay. I call back the caller, communication OK

If my provider declare his RTP [ 40000 - 41000 ]
- do you think that calling me, it will force the RTP to be at [ 40000
-41000 ] ??
my reply will be on which RTP ? [ 30000=31000 ] or [ 40000-41000 ] based
on my provider setup ?

Thanks
 
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Gordon Henderson
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      12-01-2008
In article <KlQYk.27522$(E-Mail Removed)2>,
Steve <(E-Mail Removed)> wrote:
>Hi,
>
>I configured my Asterisk server to send RTP wiht the port range : 30000
>- 31000 ]
>This way I can priotize the communication with the QoS
>
>Now, last time I received a phone call, coming from my provider, which
>I suppose got an Asterisk server as well and the COM was with a long
>delay. I call back the caller, communication OK
>
>If my provider declare his RTP [ 40000 - 41000 ]
>- do you think that calling me, it will force the RTP to be at [ 40000
>-41000 ] ??
>my reply will be on which RTP ? [ 30000=31000 ] or [ 40000-41000 ] based
>on my provider setup ?


Not sure about the "very long delay", but wouldn't it be easier to
prioritise all traffic going to/from the IP address of your asterisk
box rather than pick on specific ports?

Gordon

 
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Dave Higton
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      12-01-2008
In message <KlQYk.27522$(E-Mail Removed)2>
Steve <(E-Mail Removed)> wrote:

> Hi,
>
> I configured my Asterisk server to send RTP wiht the port range : 30000
> - 31000 ]
> This way I can priotize the communication with the QoS
>
> Now, last time I received a phone call, coming from my provider, which
> I suppose got an Asterisk server as well and the COM was with a long
> delay. I call back the caller, communication OK
>
> If my provider declare his RTP [ 40000 - 41000 ]
> - do you think that calling me, it will force the RTP to be at [ 40000
> -41000 ] ??
> my reply will be on which RTP ? [ 30000=31000 ] or [ 40000-41000 ] based
> on my provider setup ?


Audio from your provider to you could emanate from almost any source
port, but will be to a destination port in the range 30000..31000.

Audio from you to your provider could emanate from almost any source
port, but will be to a destination port in the range 40000..41000.

Dave
 
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Steve
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      12-02-2008
Gordon Henderson wrote:
> In article <KlQYk.27522$(E-Mail Removed)2>,
> Steve <(E-Mail Removed)> wrote:
>> Hi,
>>
>> I configured my Asterisk server to send RTP wiht the port range : 30000
>> - 31000 ]
>> This way I can priotize the communication with the QoS
>>
>> Now, last time I received a phone call, coming from my provider, which
>> I suppose got an Asterisk server as well and the COM was with a long
>> delay. I call back the caller, communication OK
>>
>> If my provider declare his RTP [ 40000 - 41000 ]
>> - do you think that calling me, it will force the RTP to be at [ 40000
>> -41000 ] ??
>> my reply will be on which RTP ? [ 30000=31000 ] or [ 40000-41000 ] based
>> on my provider setup ?

>
> Not sure about the "very long delay", but wouldn't it be easier to
> prioritise all traffic going to/from the IP address of your asterisk
> box rather than pick on specific ports?
>
> Gordon
>


thanks guys for your reply

Gordon, yes I thought about prioritizing the complete server, but my
fear is that if the server is trying to communicate with outside for any
reason, it will prioritize everything and will decrease the quility of
the COM.
For instance, for maintenance, I am using VNC... so the day I am
connected to it, that will will mess up the COM

Thanks
 
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alexd
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      12-02-2008
Steve wrote:

> If my provider declare his RTP [ 40000 - 41000 ]
> - do you think that calling me, it will force the RTP to be at [ 40000
> -41000 ] ??
> my reply will be on which RTP ? [ 30000=31000 ] or [ 40000-41000 ] based
> on my provider setup ?


How about 'lsof -i | grep asterisk' whilst a call is active? Should show
what ports are in use. Failing that, tcpdump will.

--
<http://ale.cx/> (AIM:troffasky) ((E-Mail Removed))
18:20:43 up 2 days, 7:01, 2 users, load average: 0.10, 0.04, 0.01
They call me titless because I have no tits

 
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Dave Higton
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      12-02-2008
In message <(E-Mail Removed)>
"Martin²" <(E-Mail Removed)> wrote:

> Sorry to hitchhike this thread, but for a long time I have been searching
> for answer to this question:
>
> How do I find which RTP ports are used by my VoIP WiFi phone (Senao 7800H)?
> I can't hear the incoming audio.


Well...

If you can get the SIP packets into Wireshark, you can see the port
numbers you're offering.

How to get WiFi packets into Wireshark is another matter. You may
be able to configure your router to copy the packets out to another
physical port and plug your Wireshark PC into that. Alternatively,
you may be able to get somebody at the other end of your call to
Wireshark the packets. Same information, different route.

Dave
 
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alexd
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Posts: n/a
 
      12-03-2008
Martin² wrote:

> Sorry to hitchhike this thread, but for a long time I have been searching
> for answer to this question:
>
> How do I find which RTP ports are used by my VoIP WiFi phone (Senao 7800H)
> ? I can't hear the incoming audio.


SIP is in theory a peer to peer protocol. So if you have a PC on the same
LAN as the Senao, you could install a softphone, dial the Senao by IP, and
run Wireshark whilst on the call and see what ports it's using.

--
<http://ale.cx/> (AIM:troffasky) ((E-Mail Removed))
21:37:44 up 2:20, 1 user, load average: 0.11, 0.08, 0.07
They call me titless because I have no tits

 
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