[SIPForum-discussion] SIPit 19 summary
Robert Sparks
rjsparks at nostrum.com
Wed Oct 25 15:30:22 UTC 2006
(This message is somewhat long, and may not survive some mail
applications attempts to be helpful with formatting.
I've also placed it at http://www.sipit.net/report19.txt)
SIPit 19 took place Oct 16-20, 2006 at the University of New
Hampshire InterOperability Laboratory (www.iol.unh.edu).
There were 140 attendees from 73 companies visiting from 16 countries
present.
There were 79 teams and 90 distinct implementations.
The Interoperability Laboratory did a spectacular job of providing a
rock-solid network for us to test on.
(For those who haven't been to a SIPit, it is a particularly intense
network torturing environment).
The majority of the spec-arguments during testing centered around how
to handle early media and early dialogs.
There was also a non-trivial subset of the implementers that were
confused about whether REGISTER and PUBLISH
create dialogs (much of this confusion seems to come from the
presence of to-tags in the 200 OK responses to
REGISTER in the examples in 3261). There were a number of interesting
questions about edge cases that I will
send to the appropriate IETF lists separately.
We tried something different for collecting data for this report at
this SIPit. We utilized a web-based
automated survey tool. As a result, we collected information on more
questions than we usually do, so this
report is a bit long. A side-effect is that the accuracy of the
information is probably a little lower. Almost
all of what's below is self-reported, and its unavoidable that for
any given question an implementer or two
didn't understand, or didn't know the answer. So, with an appropriate
level of respect for errors in sampling, here's
what we found:
The roles represented (some implementations act in more than one role):
36 endpoint
19 proxy/registrar
8 standalone proxy
4 redirect server
4 gateway
15 automaton UA (voicemail, conference, etc.)
17 b2bua/sbc
6 ua w/ signalling but no media
8 test/monitoring tool
Implementations using each transport for SIP messages:
UDP 100%
TCP 82%
TLS 45% (server auth only)
TLS 36% (server or mutual auth)
SCTP 6%
DTLS 0%
10% of the implementations supported SIP over multicast.
30% supported SIP over IPv6.
70% of the implementations correctly reassembled fragmented UDP.
Proper use of DNS for SIP continues to rise:
Full RFC3263 use of DNS 59%
SRV only 14%
A records only 15%
no DNS support 12%
Support for various items:
32% ENUM
65% rport
30% multiplexing SIP/STUN
14% SIGCOMP
25% RFC4320 fixes
14% Identity
30% connect-reuse
14 of the implementations claimed support for outbound.
Interoperability around this draft was fairly low, but the
implementers are aggressively improving it.
15 implementations claimed support for some version of GRUU. Nothing
worked together before code changes at the event. By the end a few
teams were getting scenarios to work.
Only 3 implementations were attempting to support the consent framework.
The endpoints implemented these methods:
100% INVITE and ACK
100% CANCEL
100% BYE
96% REGISTER
81% OPTIONS
76% SUBSCRIBE
80% NOTIFY
56% PRACK
52% MESSAGE
74% INFO
63% UPDATE
80% REFER
41% PUBLISH
The endpoints implemented these extensions:
67% RFC3891: replaces
63% RFC4028: session-timer
17% RFC3327: path
11% RFC3840: pref
4% RFC3841: caller-prefs
26% RFC3323: privacy
6% RFC4538: target-dialog
7% RFC4488: norefersub
56% RFC3262: 100rel
3% RFC3994: indication of message composition
44% of the endpoints implemented sipping-cc-transfer
When asked about STUN support, the client implementations replied:
6% I implement all the client requirements of draft-ietf-behave-
rfc3489bis
6% I implement some, but not all, of the client requirements of
draft-ietf-behave-rfc3498bis
13% I implement all of the client requirements of RFC3489
7% I implement some, but not all, of the client requirements of
RFC3489
59% I do not implement STUN as a client
9% Other
There are still a large number of endpoints (25%) that do not use
symmetric RTP.
There were only a couple of TURN client implementations. We had
several STUN servers and 2 TURN servers. There were only 3 ICE
implementations, and only one of those was at the current version.
Interoperability was reasonably high, but not seamless. The issues
with interoperability were all implementation problems.
I asked the endpoint implementations to characterize their handling
of S/MIME:
15% I break if someone sends me S/MIME
22% I pretend S/MIME doesn't exist if it shows up
35% I don't pay attention to S/MIME, but will proxy it or hand it
to my application
7% I pay attention to S/MIME I receive, but don't send any
2% I don't pay attention to S/MIME I receive, but I do send some
4% I try to do something useful with S/MIME I receive and send
15% Other
I asked the same question about multipart mime support:
7% I break if someone sends me multipart/mime
20% I pretend multipart/mime doesn't exist if someone sends it to me
19% I ignore multipart/mime but will proxy it or hand it to my
application if it shows up
15% I try to do something useful with multipart/mime I receive,
but I never send it
4% I ignore multipart/mime that I receive, but I try to do
something useful with multipart/mime I send
22% I try to do something useful with multipart/mime I send and
receive
13% Other
48% of the endpoint implementations claimed to correctly handle
merged requests.
Here is how the endpoints said they handled receiving 200 OKs from
more than one branch of a forked INVITE:
54% I send BYEs to all but one branch
6% I use all of them (perhaps mixing the different media streams
locally)
16% I don't handle this case gracefully
11% Other
Here is a sample of how endpoint implementors replied when asked how
they handled early media from more than one leg:
* We allow multiple RTP streams with an affinity to the last one.
* First Media received is played until 200.
* The first 183 will be honored in case of the UAC. The rest will
be dropped.
* Allow media from only negotiated address. Ignore media until
negotiated (offer-answer exchanged).
* Listen to most recently started stream.
* all early media will passed on to the UA.
* pick the one who most recently sent me a criticial threshold of
media.
* Play only one media stream and ignore others.
* The last sdp received override previous one.
* First 18x message goes through, rest dropped.
* Open voice only with the first one, but answer all of the 18x
* We will use the first recieved
* I ignore early media
* All get relayed - (all rendered leave final choice to recipient UA)
* Last early media replaces previous
* We update media as the 18x's come in. 200OK media will be the
confirmed media channel.
* Take the first
Interestingly, 15% of the endpoints supported DHCP option 120.
This is how the endpoints (that actually handled media) described
their use of RTCP:
38% I fully implement RTCP and use the RTCP from my peers
20% I send some RTCP, and open a port to receive RTCP, but I
ignore any packets I receive
18% I never send RTCP, but I do open a port for receiving (and
ignoring) it
24% I don't even open a port for RTCP
There were 12 (roughly 25%) endpoints testing SRTP support. Keying
was predominantly sdes.
Interoperability is not yet high, but more pairs got something
working than at SIPit 18.
There were only 4 endpoints supporting comedia.
There were 22 proxies present.
The proxy implementers characterized their handling of infinite loop
prevention this way:
0% I implement loop detection as per the sip-loop-fix draft
45% I perform RFC3261 loop detection
45% I don't loop detect, but do pay attention to max-forwards
10% I don't loop detect or look at max-forwards
I asked proxies "Will you proxy a request with a RURI containing an
unknown scheme
(such as splork:) when there is a Route header field whose first
value is a SIP URI
you can resolve?" and got these responses:
32% Yes
68% No
Half of the proxies in attendence actively participated in session-
timer.
There were 9 implementations (41%) that categorized themselves as
proxies that would not forward an unknown method.
Two-thirds of the proxies claimed to properly handle SIPS.
None of the proxies made use of 3840 or 3841 information
(capabilities and caller-prefs)
There were 19 registrars.
7 of the registrars (37%) accepted non-sip or sips Contacts in a
registration
11 (58%) would accept a REGISTER request that had a multipart-mime
body (almost all ignored it)
1 would accept an S/MIME signed or encrypted register
Half of the border-elements (B2BUA/SBC-like implementations) could be
configured to forward unknown methods.
75% could be configured to forward unknown SDP lines
There were 41 SIP Events implementations present
15 (37%) of them would send unsolicted notifies (there were 2 more
things that ONLY sent unsolicited notifies).
They supported these event packages:
29 refer
23 message-summary
14 presence
12 dialog
5 reg
4 ua-profile (sipping-config-framework)
4 conference
2 reg gruu extension (sipping-gruu-reg-event)
2 certificate/credentials (sip-certs)
1 session-spec-policy (sipping-policy)
1 kpml
0 vx-rtcpxr (sipping-rtcp-summary)
Only 4 (10%) supported winfo
4 supported event-list
37% of the implementation supporting SIP Events supported PUBLISH
Of the 14 implementations supporting event presence, there was
support for the following document formats:
12 base PIDF only
2 RPID
0 CIPID
0 timed presence
0 PIDF-LO
0 prescaps-ext
5 implementations supported XCAP
7 supported pres-rules
I asked all the implentations present which P- headers they actively
supported:
(I suspect many of the respondents who passively let the headers pass
or ignore them answered yes, so these
numbers, more than any others of the above are probably inflated)
28 P-Asserted-Identity
21 P-Preferred-Identity
10 P-Called-Party-ID
9 P-Associated-URI
9 P-Access-Network-Info
8 P-Charging-Vector
6 P-Visited-Network-ID
5 P-Charging-Function-Address
4 P-User-Database
4 P-DCS-* (any of the P-DCS headers)
3 P-Media-Authorization
That's it. Please let me know if there are different questions you
want to see asked next SIPit.
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