[Freeswitch-users] rtp port update after reinvite when initiator is behind nat
Sergey Safarov
s.safarov at gmail.com
Fri Jan 13 08:05:27 MSK 2017
Look this ticket FS-9206 <https://freeswitch.org/jira/browse/FS-9206>
Try this branch bugfix/FS-9206-proxy-media-with-enable-3pcc-proxy
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=refs/heads/bugfix/FS-9206-proxy-media-with-enable-3pcc-proxy>
Sergey
пт, 13 янв. 2017 г. в 1:33, <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
>
>
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