[Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true)

[Steven Ayre]

Can you supply a pastebin of adebug log of the call?

Steve on iPhone


On 12 Jul 2012, at 21:39, Adelia C. <hexade at hotmail.com> wrote:

> We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. 
> With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. 
> We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time.
> 
> Call:   
> 
> Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee.
> 
> 
> RTP streams, as negotiated (SDP):
> 
> 
> Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162  -> Callee (can't hear a thing)
> 
> Callee  -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP)
> 
>  
> 
> RTP streams, as sent/receiced:
> 
> FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080
> FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818
> FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.]
> 
> All this from Wireshark trace.
> 
> I tried the following, with no success:
> 
> 1. Added to MyFSApp.xml and dialplan\public.xml:
>     <action application="set" data="disable_rtp_auto_adjust=true"/>
>     <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> 
> Restarted, FS, same result.
> 
> 2. Commented out 
> <param name="ext-rtp-ip" value="auto-nat"/>
> <param name="ext-sip-ip" value="auto-nat"/>
> <param name="use-rtp-timer" value="true"/>
> Restarted, FS, same result.
> 
> 3. Started FS with -nonat, then nonatmap
> Same result.
> 
> Other info : 
> - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps.
> - Our FS code about 2 weeks old.
> 
> What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? 
> 
> Thanks,
> A.C.
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