[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC

Fri Jul 10 18:00:30 UTC 2020

Tried ptime 20 and 40 both on FreeSWITCH. The behavior is the same with
both. I’ll have not changed in the client, will see if there is even an
option, but this is strange behavior never seen that before.

From: Dragos Oancea <dragos at freeswitch.org> <dragos at freeswitch.org>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
<freeswitch-users at lists.freeswitch.org>
Date: July 10, 2020 at 9:36:47 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
<freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Oneway Audio, can’t send RTP back to UAC

You have ptime 40, does it work the common way (with ptime 20) ? Change it
in the client, your client makes an offer with ptime 40.
It should not be a problem (40 ms is ok too) , but maybe we get a hint on
this one way audio issue.

On Thu, Jul 9, 2020 at 9:29 PM Paul Mateer <Paul.Mateer at outlook.com> wrote:

> Don't know if it's the same problem but I incorporated FreeSWITCH code
> into a product a couple of years ago (master code was marked as 1.9.x) and
> whilst the (FreeSWITCH) server worked fine, a client built with that
> version of the software would only give audio in one direction.
>
>
>
> I figured out a fix (to switch_ivr_originate.c) and raised a Jira but
> didn't provide the fix as I wasn’t sure the change I had made was the best
> way to resolve the issue. I assumed that someone with a better
> understanding of the code would be better placed to determine the most
> appropriate fix.
>
>
>
> Interestingly if I used a much older version of the FreeSWITCH software in
> the client (like that shipping with the FSClient software) the problem did
> not occur.
>
>
>
> Sent from my Windows 10 device
>
>
>
> *From: *Muhammad Naseer Bhatti <nbhatti at gmail.com>
> *Sent: *09 July 2020 18:44
> *To: *freeswitch-users at lists.freeswitch.org
> *Subject: *[Freeswitch-users] Oneway Audio, can’t send RTP back to UAC
>
>
>
> Hi,
> I can’t seem to be able to figure out why I can’t send RTP to the other
> side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version
> 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP
> address and the other switch is also on Public IP on same subnet. I receive
> call from SippySoft (UAC) and playing delay_echo application. The call flow
> is
>
> SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP)
>
> FreeSWITCH after answering the call says
>
> 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec
> Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1]
> 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec
> Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1]
> 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec
> Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1]
> 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] is saved as a near-match
> 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting
> codec PCMU at 40i@8000h at 1c
> 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec
> sofia/internal/04232152273 at 43.225.99.130 PCMU/8000 40 ms 320 samples
> 64000 bits 1 channels
> *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111
> sofia/internal/04232152273 at 43.225.99.130 <04232152273 at 43.225.99.130>
> Original read codec set to PCMU:0*
>
> and then starts delay_echo() app. Isn’t  Original read codec set to PCMU:0
> is a bad thing? Perhaps the reason not able to send RTP back?
>
> On the other hand, I installed Asterisk 13.34.0, on the same machine, just
> to prove the point if there is network issue but things work just fine with
> Asterisk default config. Seem like there is either something not configured
> (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in
> Asterisk and FreeSWITCH both seems to be the same.
>
> FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c
> and console log is here https://pastebin.freeswitch.org/view/69b5f68c
> Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97
>
> Appreciate some input to figure out this problem.
>
>
> Thanks,
> Naseer
> _________________________________________________________________________
>
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>
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_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

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