Re: [rtcweb] draft-ibc-rtcweb-sip-websocket -- WebSocket Transport for Session Initiation Protocol (SIP)

On 09/15/2011 12:24 PM, Markus.Isomaki@nokia.com wrote:
> SIP over UDP is used by mobile operators themselves, as they are in control of the access infrastructure, and can make special provisions for their own services. But if you try to run SIP over UDP over mobile networks to any independent ("OTT") Internet service, it is likely to not work that well, due to battery consumption. Even with public addresses, in many mobile networks firewalls drop UDP flow state after 30 seconds.
>    

If the concern is the battery, then with ICE, SIP over UDP is something 
we want can reserve in stateful progressions. People don't like that 
stateful discussion when it comes to buffer size and transient mix, as 
they defaulted to cache under IPv4.

E911 traffic would be transient to corporate cache, for example. There 
are more options as we garbage collect, resolve, and reserve those 
allocations after clean from either cache or transient states. Behind 
corporate firewalls, cache items could be datestamp'd, which simply 
releases them from the transient state. Under SMTP, that was normal.

UDP is BSD friendly, and I like how rtcweb fits right in the UDP header 
with extended Content-Length:. If the SSRC is touched, then expand or 
deliver.

> TCP usually survives longer, in most networks actually more than 15 minutes. Unfortunately a small percentage of networks breaks even TCP in less than 5 minutes. (Several companies, including Nokia, have concrete stats from all over the globe.)
>
> So all in all I'm not sure that HTTP or websockets are in a worse position than pure SIP. The main differences are that the Javascript app may not have access to some helpers the more native mobile platform may offer.
>
> Markus
>
>
>    
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of ext Randell Jesup
>> Sent: 15 September, 2011 22:13
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] draft-ibc-rtcweb-sip-websocket -- WebSocket Transport
>> for Session Initiation Protocol (SIP)
>>
>> On 9/15/2011 11:57 AM, Roman Shpount wrote:
>>      
>>> Actually SIP over UDP is what is typically used for mobile apps now.
>>> If you are doing SIP from the public IP, you do not need to maintain
>>> the connection even if TCP/TLS is used.
>>>        
>> Markus had a point: SIP over UDP requires keepalives if it's behind a NAT as
>> well.  So maybe it's not such a big difference, depending on the keepalive
>> rates needed (and for UDP circa 30 sec is typical).
>>
>> --
>> Randell Jesup
>> randell-ietf@jesup.org
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>      
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    

-- 

---
<i>The wheel.</i metro-link=t dzonatasolyndra>