similar to: Asterisk server and DSCP QOS

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 ;

I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made

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

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 --

hello I''m new here can I ask about how to do a patch dscp to shorewall installed from the package in ubuntu? ------------------------------------------------------------------------------ Monitor your physical, virtual and cloud infrastructure from a single web console. Get in-depth insight into apps, servers, databases, vmware, SAP, cloud infrastructure, etc. Download 30-day Free Trial.

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

Dear all can anyone tell me how to mark packet''s DSCP value using tc? Thanks. Regards, philip -- Hong Kong IP Multicast Initiative (HKIPMI) Department of Information Engineering The Chinese University of Hong Kong Phone : 2603 5240 Fax : 2603 5032 _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl

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

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

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

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 ] >

Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091001/abf55045/attachment.htm

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

Hello list, what can I do to minimalize the jitter in SIP-calls at server level ? If at local network level, there is a VoIP-router and their is a physical network dedicated to IP-phones, but there is still jitter. When using a Hosted Asterisk server, which settings on the Asterisk-server can minimalize the jitter between the VoIP-router and the Asterisk-server on the public internet ?? Kind

https://bugzilla.mindrot.org/show_bug.cgi?id=1733 Summary: Enhance support for QoS (ToS) by supporting DSCP/CS and adding option Product: Portable OpenSSH Version: 5.4p1 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: ssh AssignedTo:

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116

Hi, I regularly have errors (kernel: dst cache overflow) and crash of a firewall under Linux 2.6.17 and the route patch from Julian Anastasov. With rtstat I see that the route cache size increases regularly without never decreasing. I have this parameters: fw:/proc/sys/net/ipv4/route# grep . * error_burst:1250 error_cost:250 gc_elasticity:15 gc_interval:60 gc_min_interval:0

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this