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    Voip : Only one way speech is working between 2 Sites!

    Scheduled Pinned Locked Moved Firewalling
    28 Posts 6 Posters 9.5k Views
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    • F
      fifinon
      last edited by

      can you explain this :

      stats table :

      udp 172.16.1.200:2910 -> 172.16.11.200:24124 -> 172.16.2.200:1719 MULTIPLE:SINGLE 
      udp 172.16.1.200:1719 -> 172.16.11.200:10490 -> 172.16.2.200:1028 SINGLE:NO_TRAFFIC

      172.16.11.200 = antenne RLAN

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      • marcellocM
        marcelloc
        last edited by

        Check if there is some nat configurarion for sip at your voip servers and also reduce  the RTP port range for a Easier rule creation.

        At asterisk its very easy to setup.

        I just don't understand why are you giving up ip phones?
        But its a firewall forum, not a voip one, So check these configs and see if ir works.

        Treinamentos de Elite: http://sys-squad.com

        Help a community developer! ;D

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        • F
          fifinon
          last edited by

          Check if there is some nat configurarion for sip at your voip servers and also reduce  the RTP port range for a Easier rule creation.

          the protocol used for my voip configuration is H323!

          can you please explain me this ( reduce  the RTP port range for a Easier rule creation )?

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          • marcellocM
            marcelloc
            last edited by

            @fifinon:

            the protocol used for my voip configuration is H323!

            can you please explain me this ( reduce  the RTP port range for a Easier rule creation )?

            I do not have experience with h323 but 'google'  ;) told me that both(sip and h323) signaling protocols uses RTP for media transport, in this case audio is the media.

            Every time you get no audio or one way audio, it means you are having RTP issues.

            At asterisk, default RTP range is from 10000 to 20000. I have no idea how h323 handles this.

            Returning to firewall….
            RTP packages sents 'inpackage' information telling other part how(and for who) he will return the package. When you have NAT, or server thinks he is behind NAT, the information inside the package will tell the other side to return the package to a wrong or unreachable destination.

            Treinamentos de Elite: http://sys-squad.com

            Help a community developer! ;D

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            • F
              fifinon
              last edited by

              @marcelloc:

              @fifinon:

              the protocol used for my voip configuration is H323!

              can you please explain me this ( reduce  the RTP port range for a Easier rule creation )?

              I do not have experience with h323 but 'google'  ;) told me that both(sip and h323) signaling protocols uses RTP for media transport, in this case audio is the media.

              Every time you get no audio or one way audio, it means you are having RTP issues.

              At asterisk, default RTP range is from 10000 to 20000. I have no idea how h323 handles this.

              Returning to firewall….
              RTP packages sents 'inpackage' information telling other part how(and for who) he will return the package. When you have NAT, or server thinks he is behind NAT, the information inside the package will tell the other side to return the package to a wrong or unreachable destination.

              maybe there is a problem in NAT ! i will wait for other idea about this because i'm newbie !
              thanks alot.

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              • F
                fifinon
                last edited by

                this is another captur of my state table in site C :

                Proto    Source -> Router -> Destination    State   
                udp 172.16.3.200:1719 <- 172.16.1.200:4562 SINGLE:MULTIPLE 
                udp 172.16.1.200:4562 -> 172.16.6.200:1719 MULTIPLE:SINGLE 
                udp 172.16.3.200:48607 <- 172.16.1.200:1719 NO_TRAFFIC:SINGLE 
                udp 172.16.1.200:1719 -> 172.16.6.200:48607 SINGLE:NO_TRAFFIC 
                udp 172.16.1.200:4561 <- 172.16.6.200:48607 NO_TRAFFIC:SINGLE 
                udp 172.16.3.200:48607 -> 192.168.24.25:56773 -> 172.16.1.200:4561 SINGLE:NO_TRAFFIC

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                • pttP
                  ptt Rebel Alliance
                  last edited by

                  Diagnostics –> Packet Capture

                  Do a "call capture" then open with Wireshark ( Telephony -> VoIP Calls -> Flow ) and check where the RTPs Come & Go, then you can figure what is happening.

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                  • F
                    fifinon
                    last edited by

                    i created  this rules but no résult

                    source (ports)  => destination (ports)

                    adresse PBX site A : 172.16.1.200 (UDP 32000-32512 ) => adresse PBX Site B 172.16.2.200 (UDP 32000-32512)

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                    • marcellocM
                      marcelloc
                      last edited by

                      Create one with serverA => serverB and serverB => serverA.

                      Free all traffic between voip servers.

                      Treinamentos de Elite: http://sys-squad.com

                      Help a community developer! ;D

                      1 Reply Last reply Reply Quote 0
                      • F
                        fifinon
                        last edited by

                        @marcelloc:

                        Create one with serverA => serverB and serverB => serverA.

                        Free all traffic between voip servers.

                        i did it !! no résult >:(

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                        • F
                          fifinon
                          last edited by

                          I think the problème is in the NAT !! because when i turn off NAT filtre in advance setup the voip work very good, but the navigation in internet don't work (no internet acces) so i don't know how to give acces to internet !!! do you have any idea ????

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                          • marcellocM
                            marcelloc
                            last edited by

                            The server is on the same subnet as machines?

                            If so, disable automatic nat and create your own nat out rules.

                            Ps.
                            It sounds strange to me that some services will need nat and some don't.

                            Treinamentos de Elite: http://sys-squad.com

                            Help a community developer! ;D

                            1 Reply Last reply Reply Quote 0
                            • F
                              fifinon
                              last edited by

                              @marcelloc:

                              The server is on the same subnet as machines?

                              If so, disable automatic nat and create your own nat out rules.

                              Ps.
                              It sounds strange to me that some services will need nat and some don't.

                              Yes the server is on the same subnet as machnies !

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                              • F
                                fifinon
                                last edited by

                                I really don't understand this problème!!!
                                i tired every thing to resolve it but no solution until now!!

                                now i'm trying to make the VOIP work just between 2 sites but the firewall still block the voip!

                                the ping between 2 sites A and B is good also the transfer of DATA from A => B and B => A!

                                Site A :172.16.1.0
                                ALCATEL PBX A : 172.16.1.200
                                Site B :172.16.2.0
                                ALCATEL PBX B : 172.16.2.200

                                i created those rules,

                                in Server A :

                                Rule 1 :

                                Lan Interface :

                                Action : Pass

                                Interface : LAN

                                Protocol : Any

                                Source  : Lan subnet

                                Destination : Single Hoste Or Aliace (Site B)

                                Geteway : default

                                Rule 2 :

                                Lan Interface :

                                Action : Pass

                                Interface : LAN

                                Protocol : TCP/UDP

                                Source  : Single Hoste Or Aliace (172.16.1.200)

                                Port : from 32000 to 32512

                                Destination : Single Hoste Or Aliace (172.16.2.200)

                                Port : from 32000 to 32512

                                Geteway : default

                                –-----------------------------------------------------------

                                in Server B :

                                Rule 1 :

                                Lan Interface :

                                Action : Pass

                                Interface : LAN

                                Protocol : Any

                                Source  : Lan subnet

                                Destination : Single Hoste Or Aliace (172.16.1.0)

                                Geteway : default

                                Rule 2 :

                                Lan Interface :

                                Action : Pass

                                Interface : LAN

                                Protocol : TCP/UDP

                                Source  : Single Hoste Or Aliace (172.16.2.200)

                                Port : from 32000 to 32512

                                Destination : Single Hoste Or Aliace (172.16.1.200)

                                Port : from 32000 to 32512

                                Geteway : default

                                –------------------------------------------------------------

                                Alcatel support say that the VOIP need just port from 32000 to 32512 but i also tried to autorise all port!! but no résult!

                                in Diagnostics: System logs: Firewall : the firewall still block the voip !!!

                                Act       Time         If            Source                Destination              Proto
                                X Oct 2 15:32:01 LAN 172.16.1.200:4489 172.16.2.200:58615 UDP
                                X Oct 2 15:31:59 LAN 172.16.1.200:4491 172.16.2.200:34195 UDP


                                Do you have any idea? ???

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                                • marcellocM
                                  marcelloc
                                  last edited by

                                  Have you disabled nat between sites?

                                  Use tcpdump at console and see packages flowing

                                  Treinamentos de Elite: http://sys-squad.com

                                  Help a community developer! ;D

                                  1 Reply Last reply Reply Quote 0
                                  • F
                                    fifinon
                                    last edited by

                                    I turned off the NAT in advance setup, it work good ! but users can't have the internet navigation!!

                                    how can i block the nat just between the 2 sites ???

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                                    • M
                                      Metu69salemi
                                      last edited by

                                      With manual outbound nat, there is two ways to do it.
                                      either you have rules to these networks with a check box: DO NOt NAT and after that destination any network with normal natting

                                      -or-

                                      almost similar, but any other destinations has to have nat rule except these 2sites.

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                                      • C
                                        cmb
                                        last edited by

                                        It's probably not NATing between the sites, it wouldn't by default at least, you would have to setup manual outbound NAT for that.

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