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

[Dragos Oancea]

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
> _________________________________________________________________________
>
> 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
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20200710/3a764768/attachment.html>