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[asterisk-users] Direct Media and message "SIP/SipAgent-00000bf9 requested media update control 26, passing it to S


 
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jonas.kellens at telen...
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PostPosted: Thu Nov 28, 2013 5:11 am    Post subject: [asterisk-users] Direct Media and message "SIP/SipAgent Reply with quote

Hello,

I have the following construction :


Provider --> SipAgent (asterisk) --> Asterisk Server_A --> IP-phone (Snom 370)

If a call comes in from the "Provider" to my SipAgent, then my SipAgent send the call to the correct Asterisk Server_A (dialplan logic based on number). The Asterisk Server_A takes the call and sends it to the IP-phone.

My SipAgent has DirectMedia=yes so there is no audio flowing through this SipAgent. It only stays in the signaling path (SIP).

My SipAgent will communicate in a SIP re-INVITE the audio ports of the Asterisk Server_A to the "Provider".
My SipAgent will communicate in a SIP re-INVITE the audio ports of the "Provider" to the Asterisk Server_A.
Audio will flow directly between "Provider" and "Asterisk Server_A".

This works great.


On my Asterisk Server_A, I see the following :

SIP/SipAgent-00000bf9 requested media update control 26, passing it to SIP/ead14-00000bfb

Mostly this appears one time in a call. This I find normal.

But sometimes the CLI is flooded with 100 of these messages... and that I find NOT NORMAL.

The flood stops when the call is anwered.



This is the SIP INVITE on my SipAgent :

INVITE sip:xx32xxxxxx@XX.XX.XX.199:5060 ([email]sip:xx32xxxxxx@XX.XX.XX.199:5060[/email]) SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.198:5060;branch=z9hG4bK37fc69a2;rport
Max-Forwards: 70
From: "xx35xxxxxx" <sip:xx35xxxxxx@XX.XX.XX.198>;tag=as3bbe54ca
To: <sip:xx32xxxxxx@XX.XX.XX.199>;tag=as180f6a04
Contact: <sip:xx35xxxxxx@XX.XX.XX.198:5060>
Call-ID: 675c1f3f5141f5ac0e981c27414de0be@XX.XX.XX.198:5060 ([email]675c1f3f5141f5ac0e981c27414de0be@XX.XX.XX.198:5060[/email])
CSeq: 103 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 239


X-Asterisk-Info shows the RTP bridge, which I find normal.

And my Asterisk Server_A answers with "100 Trying".


Now, what could be the difference between a call where the CLI on Asterisk Server_A tells requested media update control 26 one time and where it floods the CLI ?


Kind regards,

Jonas.

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