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Michael Shuler

2004-Mar-08 22:17 UTC

[Asterisk-Users] RTP Read error: Resource temporarily unavailable

I looked all over google and the mailing lists but I can't figure this out.
I can call a non NAT to a non NAT without a problem with and without
reinvite.  As soon as I try to have a call between a NAT and a non NAT I get
this...  The phones can't hear each other.  The MikeNATSnom1 has nat=yes and
is set to Automatic for NAT detection.  My test router (a Netgear FVS318)
has uPnP enable too.  From what I understood Asterisk's is supposed to be
able to deal with this.  Any thoughts?


sip0*CLI> 
    -- Executing Dial("SIP/MikeSnom1-1321",
"SIP/MikeNATSnom1") in new stack
    -- Called MikeNATSnom1
    -- SIP/MikeNATSnom1-a16c is ringing
    -- SIP/MikeNATSnom1-a16c is ringing
    -- SIP/MikeNATSnom1-a16c is ringing
    -- SIP/MikeNATSnom1-a16c answered SIP/MikeSnom1-1321
    -- Attempting native bridge of SIP/MikeSnom1-1321 and
SIP/MikeNATSnom1-a16c
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable
Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: RTP Read error:
Resource temporarily unavailable

----------------------------------------

Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: mike@bwsys.net
Customer Service: (877) 976-0711

Michael Shuler

2004-Mar-08 23:40 UTC

head link

[Asterisk-Users] RTP Read error: Resource temporarily unavailable

Did that, got the same result.  Any other ideas?  Do I need a STUN server or
does * have something built in for dealing with the RTP packets?

----------------------------------------

Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: mike@bwsys.net
Customer Service: (877) 976-0711 
> -----Original Message-----
> From: Chris A. Icide [mailto:chris@netgeeks.net] 
> Sent: Tuesday, March 09, 2004 12:20 AM
> To: Michael Shuler
> Subject: Re: [Asterisk-Users] RTP Read error: Resource 
> temporarily unavailable
> 
> 
> At 09:17 PM 3/8/2004, you wrote:
> >I looked all over google and the mailing lists but I can't 
> figure this out.
> >I can call a non NAT to a non NAT without a problem with and without
> >reinvite.  As soon as I try to have a call between a NAT and 
> a non NAT I get
> >this...  The phones can't hear each other.  The MikeNATSnom1 
> has nat=yes and
> >is set to Automatic for NAT detection.  My test router (a 
> Netgear FVS318)
> >has uPnP enable too.  From what I understood Asterisk's is 
> supposed to be
> >able to deal with this.  Any thoughts?
> 
> Mike,
> 
> Add canreinvite=no to the NAT SIP phone.  You don't want a 
> NAT'd SIP phone 
> trying to establish a Phone-phone direct link unless you know 
> exactly what 
> is going on from that phone out through the nat.  This will 
> prevent any 
> redirects from occuring when the NAT'd phone is one of the 
> clients.  The 
> media path will remain with Asterisk.
> 
> 
> >sip0*CLI>
> >     -- Executing Dial("SIP/MikeSnom1-1321", 
> "SIP/MikeNATSnom1") in new stack
> >     -- Called MikeNATSnom1
> >     -- SIP/MikeNATSnom1-a16c is ringing
> >     -- SIP/MikeNATSnom1-a16c is ringing
> >     -- SIP/MikeNATSnom1-a16c is ringing
> >     -- SIP/MikeNATSnom1-a16c answered SIP/MikeSnom1-1321
> >     -- Attempting native bridge of SIP/MikeSnom1-1321 and
> >SIP/MikeNATSnom1-a16c
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >Mar  8 19:37:29 WARNING[1234455344]: rtp.c:375 ast_rtp_read: 
> RTP Read error:
> >Resource temporarily unavailable
> >
> >----------------------------------------
> >
> >Michael Shuler, C.E.O.
> >BitWise Systems, Inc.
> >1301 W. Pioneer Parkway
> >Peoria, IL 61615
> >Office: (217) 585-0357
> >Cell: (309) 657-6365
> >Fax: (309) 213-3500
> >E-Mail: mike@bwsys.net
> >Customer Service: (877) 976-0711
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users@lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>

Rich Adamson

2004-Mar-09 05:21 UTC

head link

[Asterisk-Users] RTP Read error: Resource temporarily unavailable

> I looked all over google and the mailing lists but I can't figure this
out.
> I can call a non NAT to a non NAT without a problem with and without
> reinvite.  As soon as I try to have a call between a NAT and a non NAT I
get
> this...  The phones can't hear each other.  The MikeNATSnom1 has
nat=yes and
> is set to Automatic for NAT detection.  My test router (a Netgear FVS318)
> has uPnP enable too.  From what I understood Asterisk's is supposed to
be
> able to deal with this.  Any thoughts?
Mike,

About the only realistic way to troubleshoot nat problems like this is to
use a packet sniffer to see what addresses, etc, are actually being used on
the wire. (Download ethereal if you don't already have one.)

Part of the problem is wide variations in how well various hard/soft phones
deal with nat'ing, wide variations in the various inexpensive firewalls on
how they deal with translations (both nat and pat), variations in 
translation table timeouts in firewalls, necessary parameters needed in
* definitions for specific implementations, whether * has one or more nic
cards, etc, etc.

With the appropriate packet traces, there is a very high probability one
can make 'almost' all installations function correctly. The root issue
is
exactly how the two endpoints negotiate the rtp ports to be used for the
conversation, and a key element in that process is understanding exactly
how the nat box is impacting that rtp negotiation. You won't see enough
detail in the * sip logs (including sip debug) to ID the issue.

I think it's fair to say that in most cases involving nat, you'll end up
using "canreinvite=no" and force the rtp session through *.

After getting one phone to work from behind a nat box, there is also a
high probability placing a second call (simultaneously) from behind the
same nat box may become your next challenge. (The uPnP function adds
little if any value to the process.)

Rich

asterisk users - Mar 2004 - RTP Read error: Resource temporarily unavailable

[Asterisk-Users] RTP Read error: Resource temporarily unavailable

[Asterisk-Users] RTP Read error: Resource temporarily unavailable

[Asterisk-Users] RTP Read error: Resource temporarily unavailable