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@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@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
>
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>



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Anthony Minessale II

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