[asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short

[Alexandru Oniciuc]

Hello Wassim,

                server side you can check the RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the file isn't there, here are the defaults:

;[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000

You can even debug the RTP : CLI> rtp debug ip xxx.xxx.xxx.xxx(linksys)

Asterisk listens on one of those ports(rtp.conf ones) when a call is initiated. The same does your Linksys GW: it will listen only on the RTP configured ports.

Check the firewall between the VoIP server and the Linsys GW and check the firewall on the Asterisk server.

Debugging SIP you can see which ports are involved.

There might be other problems, maybe because you are trying to directly pass the call from one peer(let's say an external voice provider) to the other(linksys). In that case careinvite=no is be your friend.

Regards,
Alex

Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di wassim darwich
Inviato: giovedì 28 gennaio 2010 21:41
A: asterisk-users at lists.digium.com
Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short

Hi:
Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me please.


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