-------- Original-Nachricht -------- Betreff: [RAI] SIPit 28 summary Datum: Fri, 15 Apr 2011 10:47:08 -0500 Von: Robert Sparks rjsparks@nostrum.com An: rai@ietf.org
Also available at https://www.sipit.net/SIPit28_summary
RjS
--------
SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of Arpil 11-15, 2010.
There were 54 attendees from 19 companies visiting from 10 countries. We had 40 distinct implementations.
The roles represented (some implementations act in more than one role) 34 endpoints 6 proxy/registrars/b2bua/sbcs 1 dedicated event server
Implementations using each transport for SIP messages: UDP 98% TCP 95% TLS 68% server-auth, 50% mutual-auth SCTP 10% DTLS 0%
68% of the implementations present supported IPv6.
There was one RFC4474 Identity implementation present.
For DNS we had support for: Full RFC3263 : 53% SRV only : 33% A records only : 15% no DNS support : 0%
Support for various items in the endpoints: 76% replaces 38% 5389stun 26% turn 24% 3489stun 24% ice 21% sip/stun multiplexing 21% outbound 21% gruu 18% path 18% join 18% diversion 6% history-info (there were no implementations of 4244bis) 3% service-route 3% sigcomp
Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route
The endpoints and B2BUAs implemented these methods: 100% INVITE, CANCEL, ACK, BYE 94% REGISTER 91% OPTIONS 91% REFER 82% NOTIFY 82% INFO 79% UPDATE 71% SUBSCRIBE 68% PRACK 47% PUBLISH 29% MESSAGE
94% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). One of the implementations required the other party to use symmetric-rtp.
88% of the UAs present both sent RTCP and paid attention to RTCP they received.
56% of the endpoints present supported SRTP using sdes. There was one dtls-srtp implementations at this SIPit.
67% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support.
48% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 68% followed RFC4320 (corrections to the non-INVITE transaction)
There were 5 SIP Event Server implementations (not counting endpoints that support events only for refer). There were 9 SIP Event Client implementations (not counting endpoints that support events only for refer).
These event packages were supported: Server Client 2 3 presence 1 0 presence.winfo 1 6 message-summary 2 5 dialog 0 2 reg 2 1 conference 1 0 xcap-diff 0 1 kpml
These packages were supported with the eventlist extension: Server Client 1 0 presence
Five of the proxies present still rely only on max-forwards. There was one implementation of fork-loop-fix, but no implementations of max-breadth.
Multiparty tests
(Thanks to Eoin McLeod for contributing notes for several tests)
The Forking test was very well attended. We had the majority of the participants at the multiparty table. Many teams took away good new information from watching their implementation react to multiple merges and to Via stacks with transports switching between IPv4 and IPv6. Forcing multiple 200 Oks was more difficult than usual given the network equipment at hand. We've created a set of standing automated tests that will allow endpoints and proxies to test that scenario at their leisure.
The TLS multiparty leveraged DNS to reduce configuration churn. We ensured a full mesh of connections between participating elements. There were some issues uncovered with the values used in Route/Record-Route not matching the configured certificates at an element. We brainstormed a configuration for the next event that will efficiently expose whether elements are correctly opening a new connection (rather than reusing an existing connection) when a new request with a resource belonging to an authority not matching what had been authenticated on the existing connection arrived.
The SRTP tests showed a high level of interoperability (all with SDES keying), including hold/resume and transfer scenarios. There was a successful test of a call with media running long enough to increment the ROC, followed by a hold/resume. Several participants were trying variants of "best-effort/opportunistic" encryption with lower levels of success. These elements tested providing offers with both RTP/AVP and RTP/SAVP media-lines describing the same logical media channel, and with multiple RTP/AVP media lines decorated with different crypto attributes.
The STUN/TURN/ICE tests had very high levels of interoperability. The test scenario involved elements in two natted networks sharing the same address space and elements in the public address space (including a separate TURN server for each of the private network islands). We had a mix of elements that supported relay allocation and elements that presented only host/reflexive candidates. A few elements assumed that all of the m-lines in an offer had to present ICE candidates. SDP with a mix of ICE enabled m-lines and m-lines without candidates caused ICE failures.
The Outbound test showed several successful flows being created and maintained. A mesh of edge-proxies and proxy/registrars were set up and endpoints established flows through several combinations. There was good UA support for the Flow-Timer, but there were implementations present that didn't randomize the keepalive value. There was not much testing of recovery from failed flows at this event - we'll focus on that at SIPit 29.
We held a long multiparty session focusing on exchanging video with constrained bandwidth or impaired (latent or lossy) networks. There was a high level of basic interoperability and reasonable reaction to mid-call shifts in network conditions, but all parties involved came away with changes they wanted to make in their implementations.
The Spiral test presented requests to the participating UAs containing a mix of UDP, TCP, and TLS over IPv4 and IPv6 values in large Via and Record-Route stacks. Correct operation of subsequent in-dialog requests in both directions was tested, discovering a few implementation problems, and some invalid assumptions about how dialog-stateful proxies might need to behave, but most scenarios worked well. We followed the usual Spiral test with a "fuzz-ball" test utilizing DNS SRV to randomly distribute the propagation of messages through all of the proxies present, hopping off to the UAs with a low probability, resulting in an easy-to-configure test that exercised a full-mesh of connections between proxies and presented the UAs with very diverse (and large) Via and Record-Route header field values.
We exercised the RFC5393 proxy-doom scenario, again leveraging the SRV load-leveling capabilities to simplify configuration. We had messages popping out to the UAs from each level of the tree, giving them a very large-scale merge test.
_______________________________________________ RAI mailing list RAI@ietf.org https://www.ietf.org/mailman/listinfo/rai
2011/4/15 Klaus Darilion klaus.mailinglists@pernau.at:
Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route
Correct me if I'm wrong:
Items implemented in Kamailio/sip-router: - path - diversion
16 apr 2011 kl. 15.59 skrev Iñaki Baz Castillo:
2011/4/15 Klaus Darilion klaus.mailinglists@pernau.at:
Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route
Correct me if I'm wrong:
Items implemented in Kamailio/sip-router:
- path
- diversion
Yes, Kamailio is behind and after focusing so much on the merger of the code I think it would be good to plan some new features for a coming release. The issue is what of all the new SIP features we need. History-info seems useful for Asterisk, don't know how much it will affect Kamailio. Gruu, well. I haven't seen many implementations out there. I can theoretically see a need for it both in NAT situations and IPv6. Diversion is deprecated. Outbound is the final NAT traversal solution - which seems to be getting a lot of traction. Remember that there was very few participants in this SIPit so 50% is not a "market share" - far from it.
/O
2011/4/16 Olle E. Johansson oej@edvina.net:
Yes, Kamailio is behind and after focusing so much on the merger of the code I think it would be good to plan some new features for a coming release. The issue is what of all the new SIP features we need.
History-info seems useful for Asterisk, don't know how much it will affect Kamailio.
Gruu, well. I haven't seen many implementations out there.
If the registrar doesn't support GRUU then it doesn't matter whether clients support it or not :) I think GRUU can be really useful.
I can theoretically see a need for it both in NAT situations and IPv6.
Diversion is deprecated.
No, it became RFC 5806 past year:
http://tools.ietf.org/html/rfc5806
Outbound is the final NAT traversal solution - which seems to be getting a lot of traction. Remember that there was very few participants in this SIPit so 50% is not a "market share" - far from it.
2011/4/16 Iñaki Baz Castillo ibc@aliax.net:
Diversion is deprecated.
No, it became RFC 5806 past year:
But it has "Category: Historic".
Iñaki Baz Castillo writes:
No, it became RFC 5806 past year:
But it has "Category: Historic".
they could not make it anything else because official solution is history-info that almost no one has implemented.
-- juha
I can theoretically see a need for it both in NAT situations and IPv6.
Diversion is deprecated.
No, it became RFC 5806 past year:
Category: HIstoric
Because it took so much time between the Diversion draft and the SIP-history RFC, everyone implemented Diversion headers. The IETF felt that this had to be documented, but not on standard trac and not as any sort of recommendation. Result: It's still deprecated.
Not all RFCs are standard trac. The IAX2 and ZRTP are just informational RFCs even though some people claim that these protocols are now "IETF standard" since they have RFC numbers... Well, they're not. We could propably write an RFC about Kamailio's MI interface or the RTPproxy management protocol and publish, but they won't be on standard track...
/O
16 apr 2011 kl. 19.36 skrev Iñaki Baz Castillo:
2011/4/16 Olle E. Johansson oej@edvina.net:
Yes, Kamailio is behind and after focusing so much on the merger of the code I think it would be good to plan some new features for a coming release. The issue is what of all the new SIP features we need.
History-info seems useful for Asterisk, don't know how much it will affect Kamailio.
Gruu, well. I haven't seen many implementations out there.
If the registrar doesn't support GRUU then it doesn't matter whether clients support it or not :) I think GRUU can be really useful.
Me too. So maybe that's a good start.
/O
On 4/16/11 7:30 PM, Olle E. Johansson wrote:
16 apr 2011 kl. 15.59 skrev Iñaki Baz Castillo:
2011/4/15 Klaus Darilionklaus.mailinglists@pernau.at:
Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route
Correct me if I'm wrong:
Items implemented in Kamailio/sip-router:
- path
- diversion
Yes, Kamailio is behind and after focusing so much on the merger of the code I think it would be good to plan some new features for a coming release.
I think this statement is not true. While a lot of work was indeed done on merging code, there were a lot of new featured added, much more than we had we previous major releases, many based on standardizations (e.g., presence extensions to conference, dialog info, xcap, a.s.o).
The the core supports processing stun, see ser_stun.{c,h} - the guys at ser added that very long time ago.
As for next release, there are lot of features added (http://sip-router.org/wiki/features/new-in-devel), more to come, but I think most of developers here got tired hunting the ghosts of IETF specs that never were any useful nor had a touch with the reality.
As of myself, I really work now on what is demanded, there is no time to jump and implement any rfc/draft published based on theories elaborated by guys that never implemented their specs. Moreover, some of them are now in different boat and make "incredible" statements about sip considering their role in the past with this protocol...
Cheers, Daniel
The issue is what of all the new SIP features we need. History-info seems useful for Asterisk, don't know how much it will affect Kamailio. Gruu, well. I haven't seen many implementations out there. I can theoretically see a need for it both in NAT situations and IPv6. Diversion is deprecated. Outbound is the final NAT traversal solution - which seems to be getting a lot of traction. Remember that there was very few participants in this SIPit so 50% is not a "market share" - far from it.
/O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
I wholeheartedly agree with Daniel's assessment. 3.x and 3.1.x have seen incredibly important feature gains that have been revolutionary for our work. It is true that a lot of the feature growth has been in internal/runtime capabilities of Kamailio as a pseudoprogrammatic execution environment, but so what? The net benefit to implementors and users of features like configuration file splitting, new TM, rtimer, mqueue, HTTP server + XMLRPC, and many other things has been enormous.
Also, I strongly agree that the focus of the project should not be to go around chasing someone's novel, indecipherable IETF draft of the hour. They come and go like influenza. Note how many expired drafts were adopted by the market de facto immortally despite their expiration, like RPID, or, more often, completely ignored despite persistent attempts at standardisation. As Daniel said, no relation to reality in a lot of them. It becomes clear over a much longer period what should ultimately be supported, and watching the standards track is not a practical way to try to make that determination.
-- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
On Apr 16, 2011, at 2:27 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 4/16/11 7:30 PM, Olle E. Johansson wrote:
16 apr 2011 kl. 15.59 skrev Iñaki Baz Castillo:
2011/4/15 Klaus Darilionklaus.mailinglists@pernau.at:
Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route
Correct me if I'm wrong:
Items implemented in Kamailio/sip-router:
- path
- diversion
Yes, Kamailio is behind and after focusing so much on the merger of the code I think it would be good to plan some new features for a coming release.
I think this statement is not true. While a lot of work was indeed done on merging code, there were a lot of new featured added, much more than we had we previous major releases, many based on standardizations (e.g., presence extensions to conference, dialog info, xcap, a.s.o).
The the core supports processing stun, see ser_stun.{c,h} - the guys at ser added that very long time ago.
As for next release, there are lot of features added (http://sip-router.org/wiki/features/new-in-devel), more to come, but I think most of developers here got tired hunting the ghosts of IETF specs that never were any useful nor had a touch with the reality.
As of myself, I really work now on what is demanded, there is no time to jump and implement any rfc/draft published based on theories elaborated by guys that never implemented their specs. Moreover, some of them are now in different boat and make "incredible" statements about sip considering their role in the past with this protocol...
Cheers, Daniel
The issue is what of all the new SIP features we need. History-info seems useful for Asterisk, don't know how much it will affect Kamailio. Gruu, well. I haven't seen many implementations out there. I can theoretically see a need for it both in NAT situations and IPv6. Diversion is deprecated. Outbound is the final NAT traversal solution - which seems to be getting a lot of traction. Remember that there was very few participants in this SIPit so 50% is not a "market share" - far from it.
/O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla http://www.asipto.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
2011/4/16 Daniel-Constantin Mierla miconda@gmail.com:
As for next release, there are lot of features added (http://sip-router.org/wiki/features/new-in-devel), more to come, but I think most of developers here got tired hunting the ghosts of IETF specs that never were any useful nor had a touch with the reality.
As of myself, I really work now on what is demanded, there is no time to jump and implement any rfc/draft published based on theories elaborated by guys that never implemented their specs. Moreover, some of them are now in different boat and make "incredible" statements about sip considering their role in the past with this protocol...
Hi Daniel, I strongly agree with you (and sure you know I'm a good "friend" of IETF-way= :)
I just said that, IMHO, gruu and tcp-outbound are two cool features. Of course, both need implementation in client and server side. The fact that there are not gruu implementations yet doesn't mean that the feature is interesting.
Cheers.
16 apr 2011 kl. 20.27 skrev Daniel-Constantin Mierla:
On 4/16/11 7:30 PM, Olle E. Johansson wrote:
16 apr 2011 kl. 15.59 skrev Iñaki Baz Castillo:
2011/4/15 Klaus Darilionklaus.mailinglists@pernau.at:
Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route
Correct me if I'm wrong:
Items implemented in Kamailio/sip-router:
- path
- diversion
Yes, Kamailio is behind and after focusing so much on the merger of the code I think it would be good to plan some new features for a coming release.
I think this statement is not true. While a lot of work was indeed done on merging code, there were a lot of new featured added, much more than we had we previous major releases, many based on standardizations (e.g., presence extensions to conference, dialog info, xcap, a.s.o).
The the core supports processing stun, see ser_stun.{c,h} - the guys at ser added that very long time ago.
As for next release, there are lot of features added (http://sip-router.org/wiki/features/new-in-devel), more to come, but I think most of developers here got tired hunting the ghosts of IETF specs that never were any useful nor had a touch with the reality.
As of myself, I really work now on what is demanded, there is no time to jump and implement any rfc/draft published based on theories elaborated by guys that never implemented their specs. Moreover, some of them are now in different boat and make "incredible" statements about sip considering their role in the past with this protocol...
That's good feedback. I meant that we're behind the IETF. But so are most implementations out there...
Sorry if I upset anyone. That wasn't my intention.
/O