Summary: | ASTERISK-16430: TOS_SIP does not get set | ||
Reporter: | nickb (nickb) | Labels: | |
Date Opened: | 2010-07-25 22:54:36 | Date Closed: | 2010-08-23 17:02:28 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Following set in sip.conf, setting reflected in sip show settings via CLI tos_sip=CS3 tos_audio=EF Place an outbound call and capture packets on the local server RTP Packet Works: 11:39:59.554679 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) LOCAL.26392 > REMOTE.8768: UDP, length 172 Signaling Packet Broken: 11:39:59.633869 IP (tos 0x0, ttl 64, id 35957, offset 0, flags [none], proto: UDP (17), length: 479) LOCAL.sip > REMOTE.sip: SIP, length: 451 ****** ADDITIONAL INFORMATION ****** Issue can be replicated every time, tested on both 1.6.2.9 and 1.4.28. User on asterisk-users reports he can replicate it on trunk. Have tried setting it as both CS3/EF as well as 0x60/0xB8 and issue persists. | ||
Comments: | By: Philip Prindeville (pprindeville) 2010-07-26 13:29:35 I can confirm that I'm seeing this in trunk with r279118. By: Michael L. Young (elguero) 2010-08-10 12:24:51 I am using trunk on Centos 5.5 and Fedora 13 for testing. For me, it would appear to be the opposite. I see sip being tagged but rtp is not. I just wanted to add my observations on this issue. By: Digium Subversion (svnbot) 2010-08-19 16:03:23 Repository: asterisk Revision: 282893 U branches/1.4/channels/chan_sip.c ------------------------------------------------------------------------ r282893 | dvossel | 2010-08-19 16:03:22 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue ASTERISK-16430) Reported by: nickb ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=282893 By: Digium Subversion (svnbot) 2010-08-19 16:05:52 Repository: asterisk Revision: 282894 _U branches/1.6.2/ U branches/1.6.2/channels/chan_sip.c ------------------------------------------------------------------------ r282894 | dvossel | 2010-08-19 16:05:52 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue ASTERISK-16430) Reported by: nickb ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=282894 By: Digium Subversion (svnbot) 2010-08-19 16:07:18 Repository: asterisk Revision: 282895 _U branches/1.8/ U branches/1.8/channels/chan_sip.c ------------------------------------------------------------------------ r282895 | dvossel | 2010-08-19 16:07:18 -0500 (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue ASTERISK-16430) Reported by: nickb ........ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=282895 By: Digium Subversion (svnbot) 2010-08-19 16:08:37 Repository: asterisk Revision: 282896 _U trunk/ U trunk/channels/chan_sip.c ------------------------------------------------------------------------ r282896 | dvossel | 2010-08-19 16:08:37 -0500 (Thu, 19 Aug 2010) | 32 lines Merged revisions 282895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue ASTERISK-16430) Reported by: nickb ........ ................ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=282896 |