[Freeswitch-users] One Way Audio - Auto Change RTP port?

[Dan Lan]

Hi, Anthony:

Thanks for your direction. Could you give me a little bit more info about
where to set the parameters?

What I did was, I put the incoming GW IP into my ACL list, so my FS will
accept call from the GW.
I then create a public dialplan to transfer the incoming DID to a registered
SNOM phone with public IP address.
  <extension name="my_incoming_did">
    <condition field="destination_number" expression="^(999\d{10})$">
    <action application="transfer" data="$1 XML default"/>
    </condition>
  </extension>

After I add
<action application="set"  data="disable_rtp_auto_adjust=true"/>
before action "transfer"

The RTP flow become like this.
GW(5416) -->  FS (31326)
FS (31326) --> GW (5418)
This looks work fine on Leg A now without auto change the port (the incoming
leg)
However, something also change down the road on Leg B.
Now I got
SNOM(52934) --> FS (21464)
NO ANY RTP from FS --> SNOM ...

So now I still got one way voice, but is exact other way around. Before
change, the Leg B is working fine.

My question is where shoud I put
rtp_manual_rtp_bugs=accept_any_packets  ?
Do I have to put togerther this with disable_rtp_auto_adjust?

Did you just fix this problem (because you mentioned using today's git), so
I need to re-compile the most current git to fix this? (I am in window
version)

Thanks again.
Dan Lan
On Thu, Sep 8, 2011 at 2:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> variables on the leg in question
>
> disable_rtp_auto_adjust=true
>
> and/or (with today or later GIT)
>
> rtp_manual_rtp_bugs=accept_any_packets
>
>
> On Thu, Sep 8, 2011 at 3:50 PM, Dan Lan <danlanweb at gmail.com> wrote:
> > Hi,
> > I run into a weird situation. My media gateay handle voice call with 2
> > different RTP ports for send & receive
> >
> > Here is what happened. (ps: both gateway and FS are all on public IP, no
> NAT
> > involved)
> > 1. Incoming call INVITE from gateway to FS
> > Connection Information (c): IN IP4 100.100.100.100  (This is my media
> > gateway IP address)
> > Media Description, name and address (m): audio 5294 RTP/AVP 18 0 4
> > 2. FS response with session progress with media information
> > Connection Information (c): IN IP4 200.200.200.200
> > Media Description, name and address (m): audio 22428 RTP/AVP 0
> > 3. I start to see some RTP traffic exchange between FS and GW
> > from FS (22428) --> GW (5294)
> > from GW (5292) --> FS (22428)
> > please note: the GW use two DIFFERENT PORT for RTP, one for sending and
> one
> > for receiving
> > 4. For a while (about 5 secs, I think)
> > The RTP flow change on FS side to become, (there is no RTCP packet during
> > the time)
> > from FS (22428) --> GW (5292)
> > from GW (5292) --> FS (22428)
> > In other word, the FS now sending RTP to 5292 instead of 5294 (which was
> > intended in INVITE SDP message)
> >
> > And, of course, I cannot hear the voice on GW side after this.
> >
> > Anyone encounter this before? Are there any paramaters that might
> involved
> > in this auto changing RTP port behavior of FS?
> >
> > Any direction for me is appreciated, I will play around with this, and
> post
> > back my result to community.
> >
> > Dan Lan
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> >
> >
>
>
>
> --
> Anthony Minessale II
>
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