[Asterisk-Users] RTP send errors

[Michael Welter]

Michael Welter wrote:
> Michael Welter wrote:
> 
>> I'm getting the following messages when a call is answered by a SIP 
>> device:
>>
>> Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP 
>> Transmission error to 192.168.1.254:19262: Operation not permitted
>>
>> For a Cisco 7940 line, I have the following sip.conf entry:
>>
>> [desk2]
>> type=friend
>> username=desk2
>> secret=xxx
>> host=dynamic
>> dtmfmode=rfc2833
>> context=international
>> canreinvite=no
>> callerid="xxx"<3034144980>
>> mailbox=701 at xxx
>> nat=yes
>> qualify=yes
>> accountcode=xxx
>> disallow=all
>> allow=ulaw
>> allow=g729
>>
>> The Asterisk system faces the Internet on a public IP.  The phone is 
>> behind NAT.
>>
>> Asterisk version is 1.0.7.
>>
> 
> It has nothing to do with DTMF.  It is getting a few rejected rtp frames 
> from the kernel 'sendto' function immediately after the call is 
> answered.  Does anyone offer some insight?
> 
 >

After answer, a few rtp frames are being sent from sip_write to the 
NATed address of the phone (192.168.1.43) and are being rejected.  After 
that, rtp frames are correctly sent to the public address of the phone's 
firewall, and the conversation is normal.

Can anyone offer some insight?  Do I need to move to asterisk-1.2 before 
I go any further?

Thanks

-- 
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
mike at TelecomMatters.net
www.TelecomMatters.net