Questions on RTP

Discussion in 'UK VOIP' started by Steve, Dec 1, 2008.

  1. Steve

    Steve Guest

    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
    Steve, Dec 1, 2008
    #1
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  2. In article <KlQYk.27522$2>,
    Steve <> 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
    Gordon Henderson, Dec 1, 2008
    #2
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  3. Steve

    Dave Higton Guest

    In message <KlQYk.27522$2>
    Steve <> 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
    Dave Higton, Dec 1, 2008
    #3
  4. Steve

    Steve Guest

    Gordon Henderson wrote:
    > In article <KlQYk.27522$2>,
    > Steve <> 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
    Steve, Dec 2, 2008
    #4
  5. Steve

    alexd Guest

    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) ()
    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
    alexd, Dec 2, 2008
    #5
  6. Steve

    Dave Higton Guest

    In message <>
    "MartinĀ²" <> 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
    Dave Higton, Dec 2, 2008
    #6
  7. Steve

    alexd Guest

    Senao RTP ports?

    MartinĀ² 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) ()
    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
    alexd, Dec 3, 2008
    #7
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