[asterisk-users] CLI flood : requested media update control 26

Matthew Jordan mjordan at digium.com
Tue Apr 2 10:42:20 CDT 2013

On 04/02/2013 06:37 AM, Jonas Kellens wrote:
> On 04/02/2013 12:50 PM, A J Stiles wrote:
>> (Message re-ordered for readability.  The beginning is *not* the right place 
>> for your response -- answers come *after* questions, or *between* points.)
>> On Tuesday 02 April 2013, Jonas Kellens wrote:
>>> On 04/02/2013 12:35 PM, A J Stiles wrote:
>>>> On Tuesday 02 April 2013, Jonas Kellens wrote:
>>>>> Hello,
>>>>> any idea why the Asterisk CLI gets flooded by these messages ? How can
>>>>> the SIP peer /vita3/ cause this flood ?
>>>> First question:  What is "vita3" ?  A hardware SIP phone, a softphone, an
>>>> ATA or something else?
>>> The SIP peer vita3 is a realtime sip peer, installed in a hardware
>>> IP-phone (Siemens Gigaset N510 pro).
>> Have you any other Siemens Gigaset N510 pro phones in your setup?
> Yes there are. But I want to know what these messages on the CLI mean ?

The device communicating with Asterisk over SIP channel
SIP/vita3-000010af had a change in the media source (26 ==
AST_CONTROL_SRCCHANGE). This occurs when the SSRC in an RTP packet sent
by that device changed.

When in the middle of a dialling operation, we tend to log out when one
of the parties passes information to the other party. In general, this
wouldn't flood the CLI, as a party shouldn't be passing much information
off to the other parties involved in the dial.

I'm not sure why a device in the middle of a 'normal' dialling operation
(regardless of it being either the caller/peer) would switch its SSRC
rapidly in such a fashion. A pcap should show the changes in SSRC and
might illustrate what's occurring.


Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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