Looking at the pcap, there are a total of 15 packets of comfort noise. 2 of them are from the providers gateway, the remainder are from sipx to sipx.
4291, 5208, 6001, 6752, 7688, 8463, 9253, 10030, 10810, 11561, 13297, 111551, 11552 & 6741, 6743 are the ones where the itsp sends comfort noise, sipx responds by sending comfort noise to itself. I have to wonder if this is going to also have an unintended consequence of injecting noise into the otherwise fine rtp stream because its byproduct will be to inject itself into a functional stream. Even though it is looped back to itself, it becomes a processing and stream issue since the audio is anchored there. I suspect it may result in a series of garbled audio for normal conversations too (no audio problems, then a few moments of garbled audio, then fine, then when another packet is sent, it becomes garbled again for a moment). On Sat, Nov 12, 2011 at 9:10 AM, Tony Graziano <[email protected] > wrote: > I think its a sipx bug unless they convince me otherwise. Let's hope for a > JIRA and a quick fix. > > On Sat, Nov 12, 2011 at 8:15 AM, Gerald Drouillard < > [email protected]> wrote: > >> On 11/12/2011 7:14 AM, Tony Graziano wrote: >> >> packet 6752 is comfort noise, sent from sipx back to itself? i would >> think the comfort noise would need to be sent to the trunk provider. if the >> rtp keepalive loops back to sipx, its not going to keep it alive. >> >> Good catch. Moving this conversation to the dev list. >> >> >> >> On Fri, Nov 11, 2011 at 6:14 PM, Gerald Drouillard < >> [email protected]> wrote: >> >>> On 11/11/2011 5:36 PM, Tony Graziano wrote: >>> > I also decided to test it with other media services (conferencing), >>> > and it only seems to be with voicemail. So I suspect sipx needs a >>> > freeswitch tweak to get around this. Since other media services (AA, >>> > conferencing, faxetc.) have rtp going the other way, it makes me >>> > suspect its a voicemail setting (freeswitch parameter), and I recall >>> > there was a "proxy media" setting for freeswitch invoked to handle >>> > fax, so I suspect the media settings in freeswitch need to be >>> revisited. >>> Appia has no problem leaving an a voicemail over 1 min to the exact same >>> mailbox. I just find it strange that voip.ms would hang up the call >>> with media streaming in the one direction. You would think that it >>> would at least ask for a "are you still there?" response before hanging >>> up. >>> >>> A cap file for anyone interested can be found at: >>> http://www.drouillard.biz/tmp/test.cap >>> >>> >> -- >> Regards >> -------------------------------------- >> Gerald Drouillard >> Technology Architect >> Drouillard & Associates, Inc.http://www.Drouillard.biz >> >> >> _______________________________________________ >> sipx-dev mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Customers: http://myhelp.myitdepartment.net > Blog: http://blog.myitdepartment.net > > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
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