[asterisk-dev] Proposal for PJSIP TOS/COS and DSCP

[George Joseph]

On Mon, Feb 10, 2014 at 10:23 AM, Jonathan Rose <jrose at digium.com> 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.
>
>
I'd say Option 3 plus the ability to use the strings.  They remove the
ambiguity of whether a number is a DSCP or TOS value.



>
> On Mon, Feb 3, 2014 at 11:24 AM, George Joseph <
> george.joseph at fairview5.com> 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
>>
>>
>>
>>
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