[Freeswitch-users] Port changes in multiple 183s cause no audio after 200

[Brian West]

If that is what is going on, then thats a critical detail that wasn't
outlined in the JIRA, maybe we can get a new test on the latest code to
verify?

On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Just an observation, but what you wrote in the Jira appears to be
> different from your original email.
>
> 'In the debug I see where it detects the port change from 183 #1 to 183
> #2.  As such, I hear early media from both until the 200 OK.  When the call
> connects, I see no such port change in the debug, and since it's still
> listening on the wrong port (from 183 #2), there is no audio. I've seen
> some older posts where Anthony seemed against even the port change from 183
> #1 to 183 #2, yet that seems to work okay today.  It just doesn't sense the
> port change from 183 --> 200.'
>
> So it sounded like you heard both ringbacks and nothing after 200. In the
> Jira you don't talk about the 200 at all, just issues before then. Do you
> have audio working on receiving 200 now?
>
>
>
> On 16 March 2015 at 14:18, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
>
>> Michael,
>>
>> I don't believe so.  I haven't seen any activity in the Jira
>> <https://freeswitch.org/jira/browse/FS-7207>.
>>
>>
>> - Jeff
>>
>>
>> On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <
>> m.stgeorges at csl-technologies.com> wrote:
>>
>>> Steven Ayre <steveayre at ...> writes:
>>>
>>> >
>>> >
>>> > This sounds like it belongs on Jira so the issue can be tracked.
>>> >
>>> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
>>> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
>>> > Hello,
>>> > The following is on FreeSWITCH Version
>>> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
>>> 21:57:33Z 64bit).
>>> >
>>> > The design goal for this configuration is that of a simple transcoding
>>> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
>>> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
>>> there is one call flow I'm struggling with.
>>> >
>>> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
>>> signaling follows 100, 183, 200, all is well.  But if the gateway sends
>>> multiple 183s with different RTP ports, there is no audio when the call
>>> goes to 200.  See the following example:
>>> >
>>> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
>>> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
>>> signals 200 with SDP indicating audio on port 16384 (same as original
>>> 183).
>>> >
>>> > In the debug I see where it detects the port change from 183 #1 to 183
>>> #2.  As such, I hear early media from both until the 200 OK.  When the
>>> call connects, I see no such port change in the debug, and since it's
>>> still listening on the wrong port (from 183 #2), there is no audio.
>>> >
>>> > I've seen some older posts where Anthony seemed against even the port
>>> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
>>> just doesn't sense the port change from 183 --> 200.  I don't know if
>>> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
>>> >
>>> >
>>> > - Jeff
>>> >
>>> >
>>> >
>>> >
>>> ________________________________________________________________________
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>>> >
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>>> Was this ever fixed? I have a similar problem with the behavior of a
>>> Samsung PBX that connects to a FreeSwitch.
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>>
>>
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>
>
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-- 

*Brian West*
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