RTP Audio

Home » Asterisk Users » RTP Audio
Asterisk Users 2 Comments

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 host2.1.1.1
canreinvite=yes qualify=yes insecure=invite

Thoughts?

Jerry

2 thoughts on - RTP Audio

  • Is there any kind of pjsip vs old SIP (which I am using) issue happening here. (asterisk 18.14.0)

    Jerry

  • 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.