Re: RTP audio
"Joshua C. Colp" <[email protected]>
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <CAM0A2Z2j8773ZeEsRCbrGv0mVwwW75a1p_Vh1VUr+7OOPnQVCg@mail.gmail.com> |
On Tue, Oct 18, 2022 at 4:56 PM Jerry Geis <[email protected]> wrote: > > > On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis <[email protected]> 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users