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Summary:ASTERISK-10775: sip stops workking because of non rtp ports available
Reporter:Private Name (falves11)Labels:
Date Opened:2007-11-15 09:19:22.000-0600Date Closed:2011-06-07 14:02:52
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/RTP
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '12299851027@66.28.147.100' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '12297742173@66.28.147.100' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '12298283319@38.102.64.25' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '12299851027@38.102.64.25' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '13369572726@38.102.64.25' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '19417953419@66.28.147.100' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '19417953419@38.102.64.25' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.
[Nov 15 10:14:58] ERROR[30653]: chan_sip.c:16969 sip_request_call: Unable to build sip pvt data for '12078244320@38.102.64.25' (Out of memory or socket error)
[Nov 15 10:14:58] ERROR[30653]: rtp.c:2262 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call.


****** ADDITIONAL INFORMATION ******

the process is using less than 13000 file handles. It happens when when I go above 300 calls for a while. There is no crash. I compiled with no optimizations and mem_alloc debug
Comments:By: Eliel Sardanons (eliel) 2007-11-15 11:17:21.000-0600

please upload your /etc/asterisk/rtp.conf configuration file... maybe you need to extend the rtpstart-rtpend range.
By: Jason Parker (jparker) 2007-11-15 11:58:30.000-0600

Also, what is your ulimit set to?
By: Private Name (falves11) 2007-11-15 12:01:40.000-0600

root@Sipserver asterisk]# ulimit -a
core file size          (blocks, -c) unlimited
data seg size           (kbytes, -d) unlimited
file size               (blocks, -f) unlimited
pending signals                 (-i) 400000
max locked memory       (kbytes, -l) 32
max memory size         (kbytes, -m) unlimited
open files                      (-n) 400000
pipe size            (512 bytes, -p) 8
POSIX message queues     (bytes, -q) 819200
stack size              (kbytes, -s) 10240
cpu time               (seconds, -t) unlimited
max user processes              (-u) 137216
virtual memory          (kbytes, -v) unlimited
file locks                      (-x) unlimited

RTP.CONF
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=5000
rtpend=51000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
By: Joshua C. Colp (jcolp) 2007-11-19 08:54:19.000-0600

So we need to isolate the issue...

Does sip show channels show lots of hanging SIP dialogs?
By: Private Name (falves11) 2007-11-19 09:57:29.000-0600

Yes, it showed 14000 sip channels at the time. Most were showing BYE and 200.
By: Joshua C. Colp (jcolp) 2008-01-15 20:42:07.000-0600

Okay, sip history has to be enabled and a history provided for one of these dialogs. Packets may have come in out of order that screwed up dialog destruction.
By: Private Name (falves11) 2008-01-15 20:45:58.000-0600

Question from a newbie: how do I enable sip history??
By: Joshua C. Colp (jcolp) 2008-01-15 20:47:45.000-0600

sip history - Enables it
sip history off - Disables it
sip show history <dialog> - Shows history of a dialog
By: Olle Johansson (oej) 2008-01-16 10:19:45.000-0600

As there is no crash - what is the issue here?

If asterisk can't set up any more RTP channels it actually reports an error message with internal server failure. I fixed that a long time ago, and it still worked here in the training class yesterday.

Or do you mean that SIP doesn't reply at all?
By: Private Name (falves11) 2008-01-16 10:52:21.000-0600

SIP does not accept more calls. The PBX becomes non-responsive.
By: Joshua C. Colp (jcolp) 2008-02-11 12:34:02.000-0600

So did you get a SIP history of a dialog so we can see why they aren't being properly torn down?
By: Private Name (falves11) 2008-02-11 12:37:28.000-0600

I can not reproduce the error since long ago. This case should be closed.
By: Joshua C. Colp (jcolp) 2008-02-11 12:43:01.000-0600

Closed per reporter.