[cisco-voip] signaling very average bandwidth

[Jared Mauch]

On Thu, Jun 02, 2005 at 03:37:42PM +0200, Vladimir Litovka (vlitovka) wrote:
> Hello,
> 
> So, the question is - is there very very average statistics for these protocols?

	Speaking to SIP (which is my primary experience space)
the SIP messages aren't very compact at all, since they're primarily
ascii headers combined with a sdp suffix/body in most cases.

	These headers vary widely based on the SIP Proxy, SIP User-Agent,
etc..

	Some devices take a minimalist approach to these headers, others
provide lots of extra data.  Eg: a 7940/60 running a SIP image will
log more data when "call_stats: 1" is specified.

	I think it's more of a value question, with call_stats enabled,
i'm able to diagnose more of what is going on with a phone.

Rx/Dur=17,Pkt=706,Oct=112960,LatePkt=0,LostPkt=0,AvgJit=1 
Tx/Dur=17,Pkt=832,Oct=133120

	Knowing that there were no lost packets in my media is far more
important than the signaling overhead.  In most cases the signaling
doesn't need to be immediate (eg: 60ms vs 600ms) doesn't matter.

	If you're trying to get an idea, i'd look at the options
messages which are frequently used as keepalives during NAT
operations.  Here's an example:

SIP Proxy: (~445 bytes msg, not counting udp overhead)

OPTIONS sip:1101 at 120.0.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK22a3b940
From: "asterisk" <sip:asterisk at 10.10.10.10>;tag=as3352c042
To: <sip:1101 at 120.0.0.188:5060>
Contact: <sip:asterisk at 10.10.10.10>
Call-ID: 1aed267476f9b6501a2b05d929fb1c97 at 10.10.10.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 02 Jun 2005 13:57:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE
Content-Length: 0


Phone: (~668 bytes msg, not counting udp overhead)

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK22a3b940
From: "asterisk" <sip:asterisk at 10.10.10.10>;tag=as3352c042
To: <sip:1101 at 120.0.0.188:5060>;tag=000f230009a67a6605b8928e-1296d1b6
Call-ID: 1aed267476f9b6501a2b05d929fb1c97 at 10.10.10.10
Date: Thu, 02 Jun 2005 13:57:22 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER

v=0
o=Cisco-SIPUA (null) (null) IN IP4 120.0.0.188
s=SIP Call
c=IN IP4 120.0.0.188
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15





-- 
Jared Mauch  | pgp key available via finger from jared at puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.