If it helps understanding the difference between DSCP and TOS, here is a table I put together years back that shows all the common values.
QoS Values are very confusing, so I'm making a table of the different types of values that mean the same thing. USED FOR Cisco Type Prec. Level DSCP Binary DSCP Hex DSCP Decimal Diffserv/ TOS HexDiffserv/ TOS Dec. Not Used cs7 7 111XXX 40-4E Not Used cs6 6 110XXX 30-3E AUDIO ef 5 101110 2E 46 B8 184 cs5 5 101000 28 40 A0 160 af43 4 100110 26 38 98 152 af42 4 10010024 3690144 af41 4 100010 22 34 88 136 cs44100000 20 32 80 128 af33 3 011110 1E 30 78 120 af32 3 011100 1D 28 70 112 af31 3 011010 1A 26 68 104 SIP cs3 3 011000 18 24 60 96 af23 2 010110 16 22 58 88 af22 2 010100 14 20 50 80 af21 2 010010 12 18 48 72 cs2 2 010000 10 1640 64 af13 1 001110 0E 14 38 56 af12 1 001100 0D 12 30 48 af11 1 001010 0A 10 28 40 cs1 1 001000 08 8 20 32 DATA be 0 000000 00 0 0 0 Precedence Values: 7 Reserved 6 Routing 5 Voice 4 Streaming Video 3 Call Signalling 2 Transactional 1 Bulk Data 0 Best Effort On Mon, Feb 10, 2014 at 11:23 AM, Jonathan Rose <[email protected]> wrote: > I've been looking into this issue and researching how TOS became the > DiffServ field, but there are some issues with changing configuration > options during a release that we feel the community needs to have some > input on. We have a few options for going forward here, and some are more > disruptive to existing configurations than others. > > 1. We can change the names of the fields and correct the behavior so that > all values set for pjsip.conf are DSCP values. In this case, 'tos' becomes > 'dscp' while 'tos_audio' and 'tos_video' become 'dscp_audio' and > 'dscp_video' respectively. Databases and the alembic scripts we supply for > building them would need to add the new fields. This change would certainly > break existing configurations if we don't alias the TOS fields. If we do > alias the TOS fields, then people expecting the current behavior will > likely have QOS values that don't reflect their expectations. It's an > obscure problem that would probably go unnoticed for a while. > > 2. We can have the 'tos' value for transports be set correctly by dropping > the ECN portion (bits 7 and 8) and have the remaining 6 bits form the DSCP > value so that it is being set as a TOS value would be. The benefit here is > that this would require no significant changes on the configuration end > while still being more or less accurate to what the settings are. We would > probably want to display a warning message indicating that the field > doesn't have support for the ECN bits if a value is supplied that includes > them. In this scenario, 'tos_audio' and 'tos_video' for endpoints could be > left alone and remain functioning as they do in chan_sip. > > 3. We can leave the existing TOS values functioning in the TOS manner > while adding DSCP values that override them if provided. This would require > correcting the behavior of the 'tos' setting for transports while we leave > 'tos_audio' and 'tos_video' in 12 and then adding options for 'dscp', > 'dscp_audio', and 'dscp_video' to trunk. In trunk the TOS options would be > considered deprecated (but still functional). This is probably the cleanest > way to do it, but it's still a bit more disruptive (when Asterisk 13 gets > rolled out anyway) than option 2. > > Another thing to consider is whether or not we want to add support for > reading the TOS/DSCP values as strings. While this wouldn't break any > existing pjsip.conf files, we would need to update the alembic scripts for > changing the TOS values to be strings in databases and this would require a > little effort in upgrades. This is another change that might be better left > as trunk only. > > Personally, I favor options 2 and 3. If the community feels differently > though, I'm open to other approaches. > > > On Mon, Feb 3, 2014 at 11:24 AM, George Joseph < > [email protected]> wrote: > >> >> As I was playing around with TOS/COS in pjsip last week I noticed some >> inconsistencies that I'd like to correct... >> >> There's a 'tos' parameter on the transport object but not only does it >> actually set DSCP instead of TOS, it sets it in pjproject only and >> therefore for signalling only. >> >> There are tos_audio and tos_video parameters on endpoint which do set tos >> for the rtp engine but they don't accept hex or symbolic values as chan_sip >> does. >> >> So, given that DSCP has been around for a while and that's what the >> symbolic values represent anyway, Id like to do the following... >> >> In transport, rename tos and cos to dscp_sip and cos_sip respectively and >> for dscp_sip, create a new function ast_str2dscp() as a companion to the >> existing ast_str2tos that resolves the symbolics. The new function is >> needed because while the existing ast_str2tos function takes in a string >> representation of DSCP, it left-shifts it 2 bits to go into the >> rtp_engine's tos field. pjproject only accepts dscp. >> >> In endpoint, rename tos_audio and tos_video to dscp_audio and dscp_video >> respectively, then use ast_str2tos to resolve the symbolics. >> >> I could also add the new parameters and leave the old ones there for >> backwards compatibility for a while if that makes sense. >> >> What do you think? >> https://issues.asterisk.org/jira/browse/ASTERISK-23235 >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-dev >> > > > > -- > Jonathan R. Rose > Digium, Inc. | Software Engineer > > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct +1 256 428 6139 > > Check us out at: http://digium.com & http://asterisk.org > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org
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