[OpenSIPS-Users] Message compression in OpenSIPS 1.12

[Saúl Ibarra Corretgé]

Hi Razvan,

On 02 Sep 2014, at 18:25, Răzvan Crainea <razvan at opensips.org> wrote:

> Hi all,
> 
> The first topic presented in the last public meeting[1] was related to Message compression - the idea is to reduce the size of the SIP messages to avoid UDP fragmentationor to save bandwidth and processing power.
> 
> This topic was split in two parts:
> 
> 1) SIP-wise changes over the messages (RFC3261 compliant -> supported by all user agents)
>    a) function that converts all header names to the short form (i.e. From -> f, Call-ID -> i, etc.). A single function will be exported in the script that will be called for the requests that should use the short form.
>    b) function that joins together headers that support multiple bodies (i.e. Record-Route, Via, Contact, etc.). There will be a function that indicates whether the outgoing request should merge these headers in a single one. An example of usage is when having two Via headers such as the following example:
> 
> Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
> Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
> 
> These headers will be merged in a single header:
> 
>    Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1, SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
> 
>    c)delete irrelevant headers: create a new function that is used to delete certain headers that are not mandatory in the SIP protocol. By default, all headers but the mandatory ones are deleted. However, the function should receive a headers whitelist as a parameter: a list of headers that should not be deleted because they might be useful for the platform or end-users.
> 
>    comp_delete_headers("X-Source-IP|User-Agent|Date");
> 
>    d) remove redefined SDP attributes: provide a function that removes attributes that are redefined for some of the static codecs. For example, consider the following SDP:
> 
>    m=audio 49230 RTP/AVP 0 98
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:98 L16/11025/2
> 
> The SDP above can be suppressed to the following form:
> 
>    m=audio 49230 RTP/AVP 0 98
>    a=rtpmap:98 L16/11025/2
> 
> 

I think this is a good idea. I actually have ran into this not so long ago :-) Of course the solution would be to switch to TCP, but well, not everyone can do that...


> 2)Compressing the SIP message (using gzip). The idea is to take the SDP body and several headers that are not used in the routing logic, compress them, apply a base64 transformation and add to the message's body. A use case for this is a platform that has several edge servers (SBCs) and a few core instances - when entering the platform the message compression should be applied and then sent to the core servers. Inside the core networks, the messages should be carried in the compressed format to reduce the bandwidth. When leaving the network, the message has to be decompressed and forwarded to the next gateway without any compression, since the other equipments might not understand them.
> There will be several functions exported in the script:
> 
>    a) compress_msg("1","Header1|Header2"); compresses the body of the message and listed headers
>    b) decompress_msg(); decompress both headers and body
> 
> What do you think about this approach? Is this something you find useful? Since we don't have a final decision for this topic, we are looking for more input from you guys.Anybody is welcome to throw any kind of useful feedback on this matter, so don't be shy!
> 


IIRC this is not standard, and Apple uses it somewhere on their FaceTime implementation. Kamailio has it and someone was working on a patch for PJSIP, but other than that I’m not sure how useful it is, servers could use TCP between them.


Regards,

--
Saúl Ibarra Corretgé
AG Projects



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