similar to: RTP/SIP traffic prioritization and Linux issues

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm investigating the possibility of using Asterisk as much for internal communication in an office as between offices and I would like to know what considerations you could comment to me being based on the experience that you have had. A priori two things come to my mind: * As to network topology, is advisable to have switches and

Hello - What I want to do seems very simple - I want to make sure any H.323 traffic gets processed before anything else entering or leaving this network. The network has a videoconferencing device on the LAN at 192.168.16.4. A Linux firewall NATs an external IP Address to this internal address and I have appropriate SNAT and DNAT rules that work. The NAT and connection tracking rules all work

Hi, I am trying to get assured forwarding/expedited forwarding with gred and htb working. Below is the script I am using. The following steps are what I thing is how the script works. My problem is that if I remove the HTB qdisc from the script and have the GREDS parent as the dsmark it works, but when I add the htb as a parent of GRED and DSmark the parent of htb it does not work? Any

I am assuming that the TOS values directly map to DSCP values in the ip header. Is this a correct assumption? If so, can someone tell me the correct setting to set call control packets with a DSCP of AF31(011010) and media with EF(101110)? So would the setting for AF be TOS=46?? Is it possible to mark the media and call control separately?? -------------- next part -------------- An HTML

I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184

Real networks use either PREC (as it maps 1:1 to 802.1p and MPLS TC) or DSCP. Interactive SSH uses PREC 0x0, which is just best-effort and DSCP 0x4 which has no standard meaning (found network where DSCP 0x4 was dropped, completely, as it didn't hit any defined/allowed QoS class, obviously misconfig, BE class should eat anything not already defined) Should interactive use TOS value which has

I''m successfully using HTB + GRED to shape traffic based on the DSCP field. I would like to strip the DSCP and possibly replace it with normal ToS bits on egress traffic leaving my network. Leaving DSCP set is pointless, and could potentially cause problems with some ISPs that use DSCP internally I suppose. Setting ToS bits would seem ideal as most networks still honor it to varying

honored by DSCP (first 5 bits)- even old equip should be DSCP "compatible"...or I need to do more reading :) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

https://bugzilla.mindrot.org/show_bug.cgi?id=1964 Bug #: 1964 Summary: QoS/DSCP names false translated to ToS hex value Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: amd64 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh

Hi, Can anybody tell me how to delete DSCP or TOS setting using iptable command. iptables --list OUTPUT --table mangle Chain OUTPUT (policy ACCEPT) target prot opt source destination DSCP tcp -- anywhere anywhere tcp spt:http DSCP s et 0x08 DSCP udp --

Hi, I am working on voice application. I want to identify RTP packets and set DSCP for those. Is there anyway to accomplish this task either using tc or iptables. Please help me.. Thank you -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

Hello everybody, I have tried several iptables versions (from 1.2.4 to 1.2.6a) to mark outgoing packets (change the DSCP field) with no success. #iptables -A OUTPUT -t mangle -d 10.0.0.3 -j DSCP --set-dscp 0x2 iptables v1.2.6a: unknown arg ''--set-dscp'' TOS target is working fine, but I want to create different codes for EF and AF classes in a diffserv

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

This problem is driving nuts, so I am seeking help here. Your help will be deeply appreciated. I have made myself a Linux bridge with eth1 and eth0 to form br0. Then I run a script to configure tc with htb on it. But I can never match non-icmp traffic ( such as tcp and udp ) with TOS or DSCP values such as 0x68. The full story as follows :- 1. On the source testing machine, I do this to set

Did anyone ever answer this one? THIS is what I am trying to do: >[LARTC] cbq+sfq and DSCP marking >Maria Joana Urbano stmaria@dei.uc.pt >Thu, 13 Feb 2003 19:29:42 +0000 > > * Previous message: [LARTC] Monitoring.... > * Next message: [LARTC] two routes 1 network card > * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] >

Hi, DSCP match in /tcrules/ doesn''t work with CS6 and CS7, it provides an error "invalid value" for string and hexa values. It seems that it comes from /Chain.pm/, in the function /do_dscp/: fatal_error( "Invalid DSCP ($dscp)" ) unless defined $value && $value < 0x2f && ! ( $value & 1 ); I dont understand why "$value < 0x2f", but

Hi all, I am wondering if anyone can help me to resolve a problem. I am trying to use tc command in linux to drop udp packets of specific diffserv value. I am able set diffserv value successfully in the udp packet using command:- [root@scotch src]#iptables --table mangle --append OUTPUT \ --out-interface eth0 --protocol udp --source-port 5060 \ --jump DSCP --set-dscp 8 but i am not able to

Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [root at pbx01 asterisk-1.6.1.1]# grep -i speex