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

David Villasmil david.villasmil.work at gmail.com
Sat Jul 11 02:57:08 UTC 2020


Can you share a complete sip trace? And fs Sofia config?

On Fri, 10 Jul 2020 at 19:30, Muhammad Naseer Bhatti <nbhatti at gmail.com>
wrote:

> 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
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
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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
>
> 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
>
> _________________________________________________________________________
>
> 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

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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