[asterisk-users] RTP audio

[Joshua C. Colp]

On Tue, Oct 18, 2022 at 4:56 PM Jerry Geis <jerry.geis at gmail.com> wrote:

>
>
> On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis <jerry.geis at gmail.com> wrote:
>
>> Has there been issues where "once in a while" RTP audio does not work ?
>>
>> Example: connection to Cisco call manager - works mostly all the time.
>>
>> once in a great while - person does not hear the "beep" when calling in.
>> once in a great while - person they hear the beep - but do not hear the
>> audio public address.
>>
>> What would I be looking for to track this beast down ?
>>
>> This is my SIP trunk
>> [LSVOIP]
>> type=friend
>> dtmfmode=rfc2833
>> secret=password
>> username=LSVOIP
>> defaultuser=LSVOIP
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> context=incoming
>> host=172.1.1.1
>> canreinvite=yes
>> qualify=yes
>> insecure=invite
>>
>> Thoughts?
>>
>> Jerry
>>
>
>
> Is there any kind of pjsip vs old SIP (which I am using) issue happening
> here. (asterisk 18.14.0)
>

No. The media stack between the two is the same, and is the existing one
that has existed for years. The starting point for any issue like this is a
packet capture that you can examine in wireshark to see what media is
flowing, if any, where, and the signaling.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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