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Summary:ASTERISK-09932: asterisk doesn't respect rtp port range
Reporter:pj (pj)Labels:
Date Opened:2007-07-23 06:19:46Date Closed:2011-06-07 14:07:52
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/RTP
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) rtp.conf
( 1) sip_debug1-bind_error.txt
( 2) sip_debug2-bridge_failed.txt
( 3) sip_debug3-rtp_port_1620.txt
Description:my firewall permits port range 10000-20000 for rtp passhrough (as set in rtp.conf), it worked until todays update, seems, that port range now isn't respected, because  firewall is dropping packets to asterisk server on random high port numbers and audio doesn't passthrough
when I permit all ports >1023 to asterisk, it works again.
SVN-trunk-r75505 is OK
SVN-trunk-r76467 FAIL
Comments:By: pj (pj) 2007-07-23 06:25:56

client: 192.168.164.154
asterisk: 192.168.38.20

kernel: [CHAIN INPUT] IN=eth0  SRC=192.168.164.154 DST=192.168.38.20 LEN=200 TOS=0x02 PREC=0x00 TTL=58 ID=6664 PROTO=UDP SPT=11512 DPT=9854 LEN=180
kernel: [CHAIN INPUT] IN=eth0  SRC=192.168.164.154 DST=192.168.38.20 LEN=200 TOS=0x02 PREC=0x00 TTL=58 ID=6764 PROTO=UDP SPT=11512 DPT=9854 LEN=180
kernel: [CHAIN INPUT] IN=eth0  SRC=192.168.164.154 DST=192.168.38.20 LEN=200 TOS=0x02 PREC=0x00 TTL=58 ID=6826 PROTO=UDP SPT=11512 DPT=9854 LEN=180
kernel: [CHAIN INPUT] IN=eth0  SRC=192.168.164.154 DST=192.168.38.20 LEN=200 TOS=0x02 PREC=0x00 TTL=58 ID=6890 PROTO=UDP SPT=11512 DPT=9854 LEN=180
kernel: [CHAIN INPUT] IN=eth0  SRC=192.168.164.154 DST=192.168.38.20 LEN=200 TOS=0x02 PREC=0x00 TTL=58 ID=6950 PROTO=UDP SPT=11512 DPT=9854 LEN=180
By: Brett Bryant (bbryant) 2007-07-23 13:05:31

pj, can you post your rtp.conf config? I don't see anything obvious in the code, and not much has changed between those revisions.
By: Joshua C. Colp (jcolp) 2007-07-23 13:35:20

Please also provide a sip debug so we can confirm the ports in it.
By: Donny Kavanagh (donnyk) 2007-07-23 13:37:11

i dont know if its related at all, but someone posted this into #asterisk-bugs last night. '<slavon_net> hello all. In last svn-trunk (r76484) Contact sip header always have port "0". Example: Contact: "<sip:asterisk@87.255.0.131:0>". Thanks.'
By: pj (pj) 2007-07-23 14:51:15

uploaded three sip debugs, not only wrong rtp audio port (debug3), but also another weird errors occured (Unexpected bind error, or sip-iax Bridge failed).
all three attempts with exactly same settings, no change at asterisk or client side. next attempt can be successfull (or not), errors are random.
problem with wrong port numbers happened also from skinny phone.
By: pj (pj) 2007-07-30 04:23:02

seems, that issue was transient, in current trunk r77631 it doesn't appear, but please let this tiket open, I will report after some days, if problem is definitively gone away.


By: Jason Parker (jparker) 2007-08-21 11:51:02

pj, any update here?
By: pj (pj) 2007-08-21 12:54:23

it's OK (SVN-trunk-r79999M), so, please close this bugreport. thanks
By: Joshua C. Colp (jcolp) 2007-08-27 10:39:14

Closed per request.