[Freeswitch-users] rtp port update after reinvite when initiator is behind nat

[lionelfont at free.fr]

Hi,

i'm an user from freeswitch because my trunk's provider is using a freeswitch.

i can't give you the configuration using by my provider and i can't give you any trace from freeswitch

but i'd like explain my problem to know if anyone have an idea from what can cause my issue.

My situation:

       Freeswitch (public ip A.A.A.A) <----> WAN <---> (my public ip B.B.B.B) FIREWALL <----> (my private ip C.C.C.C) IPBX ALCATEL

       During a call, after invite or reinvite, Freeswitch send me RTP DATA on wrong RTP port... 

       For sample:
           I receive a call, after invite from FS and OK from my IPBX, we have a communication who is good. We hear on the two sides.
               FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ---------------- C.C.C.C:RTPC1 <> IPBX

           If i need to hold the call, all is ok. My holding music is hearing.
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ---------------- C.C.C.C:RTPC1 <> IPBX
           FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE)
          (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX


           But when i unhold the call, "randomly", FS send me RTP DATA on the RTP port used before the unhold.. And we have no sound on the two sides...  

           Case good:
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX
           FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE)
          (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ---------------- C.C.C.C:RTPC3 <> IPBX
        

           Case bad:
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX
           FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE)
          (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX
           FS<-- A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ---------------- C.C.C.C:RTPC3 <-- IPBX
           FS--> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 --> IPBX


I don't understand why the rtp port is not good updating by freeswitch randomly. I know freeswitch don't use the port that my IPBX give in INVITE packet because it's not the good with the nat from my firewall. I suppose that's the "NDLB-connectile-dysfunction" who is used by freeswitch to find the good rtp port.

If with a little congestion, Freeswitch receive after my reinvite, one or two RTP packet from previous RTP DATA, Freeswitch can use the wrong RTP port? Is there a parameter to ignore after reinvite the five first RTP packet to autoadjust the RTP port?

Thanks to all helping and sorry for my poor english.

Best regards