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

[Gavin Henry]

Hi Dan,

We're currently debugging this also with Snom 3XX/8XX on Draytek 2820 routers:

http://jira.freeswitch.org/browse/FS-3552

On 9 September 2011 00:17, Dan Lan <danlanweb at gmail.com> wrote:
> 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
>> >
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>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
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