Re: [Asterisk-Users] Spring VON Wrap Up
On 05-04 14:35, Steven Sokol wrote: TCP/TLS would be used for the SIP messaging which handles call setup, teardown, and other non-Realtime functions. The voice stream will still be handled via RTP which is a UDP-based protocol. The reason for doing the call setup as TCP is to allow for TLS encryption. The SIP messages themselves are simply bits of ASCII text (much like SMTP messages). Currently Asterisk does SIP over UDP only (I think...). In order to support SIPS (Secure SIP, like HTTPS) we need to build a version of chan_sip (or chan_sip2 ;-) that supports SIP over TCP. The voice stream will remain UDP an therefore not succumb to enormous delay. There are some more reasons -- transport of big SIP messages and avoiding network congestion among them. SIP message can get pretty big when XML encoded documents (presence documents, for example) are attached. TCP does not fit everywhere. It is still advantageous to let SIP phones use UDP when communicating with a proxy because the proxy does not have to keep a list of opened connections which is very resource consuming (just consider that you have 10 users using the same proxy -- that can be easily achieved using single server). On the other hand, TCP is useful for proxy-to-proxy communication, especially when there is bigger amount of traffic between proxies. In this case TCP head blocking is really not a problem because the sender gets constant feedback from the remote party and can retransmit the lost segment in a short time. (There was a technical report on this published by Henning Schulzrinne). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spring VON Wrap Up
Having just returned from four days at the VON show in Santa Clara, I thought I would submit a highlights message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. [VoIP IS BIG] First, I have to say that VoIP is BIG. It is the buzz technology of the day. The show was packed, and everybody there was there for a reason. Jeff Pulver, in his introductory remarks told us the walking dead count was zero and he was right. Wall-to-wall VoIP people. [Who Was There] The crowd was a mix of service providers (including CLECs, VoIP pure-plays, ISPs adding VoIP as a service, etc.) and VoIP product vendors looking to sell solutions to the providers. Also sprinkled into the group were regulators from the FCC, advocates for various technologies, representatives from various industry groups, and a fair number of lawyers. Perhaps the most interesting story here was the nearly even split between US citizens and those from other nations. [What Was Hot] 1. SIP. Every presentation I saw mentioned SIP at some point. While it has been obvious for some time that SIP is poised to become _the_ standard for telecom in the this century, the constant repetition is a good indicator that the standards wars are actually over and SIP stands as the survivor. 2. Presence. Everybody wants to know when and where everybody is at all times. Buddy lists are in, dial-pads are out. The message is also clear that presence will go beyond online/away/offline to include actual geographic location. It will also move away from device-centric presence (knowing that a cell phone is on) to user-centric presence (knowing how a user wants to communicate at the time). We need to add presence to Asterisk. Now. 3. Asterisk. While those of us in the Asterisk community have known for some time that Asterisk can do nearly anything, given a bit of time and effort, the word seems to have spread. Asterisk was mentioned in Keynotes, Industry Perspectives, the Town Hall meeting, and in numerous breakout sessions. Hundreds of people came by the Digium/Asterisk booth to either find out more about the system, or to crow about what they are doing with Asterisk. In a feat of irony worthy of mention, Pingtel announced their new SIP Forge organization over an audio conference hosted on an Asterisk system. Asterisk is definitely hot. 4. EoIP (Everything Over IP). The lingo of the trade seems to be changing as things mature. Voice is just one application among many. Robert Pepper of the FCC described that agency's focus as moving to IP communications in general, rather than simply Voice. This makes sense. Voice really _is_ just one of many modes of communication, and a long way away from the original VoIP service. 5. Regulatory Concerns. Several of the presenters brought up social an legal issues related to VoIP, and the associated government regulations that follow. E911 service and CALEA (wiretapping) were both the big concerns, as was inter-carrier compensation and taxation. Dr. Pepper indicated that he was pleased with the direction that the VoIP market is going, in terms of the voluntary compliance with the relevant rules from the existing PSTN regs. He indicated that the FCC was, for the time being, willing to regulate minimally -- following the same model used for the Wireless carriers over the past decade. 6. VoIP Broadband Services. With ATT's announcement that it was moving into the residential and business VoIP market (joining Packet8, Vonage, and countless others), it became clear that the industry has moved beyond how to do VoIP, and into the era of how make money at VoIP. This is a fantastic change for everybody, including the Asterisk community. The gold rush has started, and those of us who understand Asterisk are in a great position to sell shovels to those heading west. Many CLECs and ISPs moving into the business are in need of solutions that work and people who can configure them. Do the math. 7. Session Border Controllers. Everybody seems to want to build walled gardens at this point. Some to keep customers from ENUMing their way to no-cost phone service, others to keep potential bad guys from abusing their resources. Nearly every presentation (at least the technical presentations) mentioned SBCs and the associated positive and negative effects they have on VoIP adoption and scalability. The jury is still out on whether the net result is positive or negative. Thoughts? [Thanks To Digium] Digium's booth became the home-away-from-home for the Asterisk community. At times there were probably 20 to 30 people crowded in and around the display. Many thanks to Mark and Greg who let all of us gather and (I hope) help pitch Asterisk and Digium. [Retraction (Steve Eats Crow)] I would like to retract a statement I made in an earlier report from the show. After sitting through two presentations by ATT, both pitching their new CallVantage
Re: [Asterisk-Users] Spring VON Wrap Up
Steven Sokol wrote: Having just returned from four days at the VON show in Santa Clara, I thought I would submit a highlights message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. Thank you for a good report! Comments inline: 1. SIP. Every presentation I saw mentioned SIP at some point. While it has been obvious for some time that SIP is poised to become _the_ standard for telecom in the this century, the constant repetition is a good indicator that the standards wars are actually over and SIP stands as the survivor. No one mentioned H.323 any more. It's SIP and only SIP. 2. Presence. Everybody wants to know when and where everybody is at all user wants to communicate at the time). We need to add presence to Asterisk. Now. Right. In SIP and IAX2. Maybe see if we can use Jabber/XMPP for IM integration. 3. Asterisk. While those of us in the Asterisk community have known for SIP Forge organization over an audio conference hosted on an Asterisk system. Asterisk is definitely hot. SIPfoundry.org - no source available yet. And yes, they showed a lot of interest to cooperate with Digium and the asterisk.org community. 4. EoIP (Everything Over IP). The lingo of the trade seems to be changing as things mature. Voice is just one application among many. Robert Pepper of the FCC described that agency's focus as moving to IP communications in general, rather than simply Voice. This makes sense. Voice really _is_ just one of many modes of communication, and a long way away from the original VoIP service. Asterisk SIP supports video now. We're a multimedia platform. 5. Regulatory Concerns. Several of the presenters brought up social an legal issues related to VoIP, and the associated government regulations that follow. E911 service and CALEA (wiretapping) were both the big concerns, as was inter-carrier compensation and taxation. Dr. Pepper indicated that he was pleased with the direction that the VoIP market is going, in terms of the voluntary compliance with the relevant rules from the existing PSTN regs. He indicated that the FCC was, for the time being, willing to regulate minimally -- following the same model used for the Wireless carriers over the past decade. Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. [Asterisk Get-Together] About 25 of us (I think) gathered at the Mexicali Grill in Santa Clara for a post-show celebration and discussion. It was a BLAST. Even as tired as most of us were (four days of trade show can wear down just about anybody) we all had a great time. It was cool to be able to put faces with names/email addresses. I think Olle Johansson took pictures of the event. They may already be on the WiKi in fact. Not yet, but I'm working on getting them uploaded. Still trying to get accustomed to the cold weather and strange time zone up here in the north. Thank you Steve for organizing this meeting! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? TCP/TLS would be used for the SIP messaging which handles call setup, teardown, and other non-Realtime functions. The voice stream will still be handled via RTP which is a UDP-based protocol. The reason for doing the call setup as TCP is to allow for TLS encryption. The SIP messages themselves are simply bits of ASCII text (much like SMTP messages). Currently Asterisk does SIP over UDP only (I think...). In order to support SIPS (Secure SIP, like HTTPS) we need to build a version of chan_sip (or chan_sip2 ;-) that supports SIP over TCP. The voice stream will remain UDP an therefore not succumb to enormous delay. -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:34 PM, James Golovich wrote: On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. Ah, okay. That makes sense. Thanks. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Scott Laird wrote: On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote: Members of the IETF added information on the to-be-standardized standard, meaning that SIP with TLS over TCP will be mandatory. We need to start working on TCP and TLS support. Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? SIP over TCP means signalling over TCP. Media is still usually RTP/UDP. SIP over TCP and TLS authenticates both ends and may also protect the signalling with encryption. SRTP protects RTP/UDP media with encryption. There are concerns that sending positioning within SIP/UDP will reveal private detailes, like position. Hence the encryption requirement. The position data needs to be given by the ISP in DHCP configuration. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
James Golovich wrote: On Mon, 5 Apr 2004, Scott Laird wrote: Could someone explain to me why anyone in their right mind would ever want to run VoIP (or any lossy real-time data) over TCP? Unless I'm missing something, the effects of packet loss would be almost perfectly pessimal. Every time you lose a packet, the receiver stalls and then can't catch up, so you get horrifically huge delays. Does it actually gain something for anyone doing voice or video? The RTP would still be UDP. Just the SIP part (call signaling) would be TCP. SIP can be TCP or UDP, many implementations (including asterisk) support only UDP. TCP for SIP (especially with TLS) will reduce the risk of a mitm attack. ...and SIP over TCP is a requirement in the SIP RFC... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Steven Sokol wrote: I think Olle Johansson took pictures of the event. They may already be on the WiKi in fact. I've uploaded the pictures without editing at http://www.voip-forum.com/asterisk/von2004/index.htm Enjoy! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote: SRTP protects RTP/UDP media with encryption. There are concerns that sending positioning within SIP/UDP will reveal private detailes, like position. Hence the encryption requirement. The position data needs to be given by the ISP in DHCP configuration. This brings up two more questions: 1. What does 'positioning' mean in a SIP context--Google isn't helpful. Is this basically just physical location? 2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients out there? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spring VON Wrap Up
K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Monday, April 05, 2004 3:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Spring VON Wrap Up Steven Sokol wrote: I think Olle Johansson took pictures of the event. They may already be on the WiKi in fact. I've uploaded the pictures without editing at http://www.voip-forum.com/asterisk/von2004/index.htm Enjoy! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
At 01:44 PM 4/5/2004, you wrote: Having just returned from four days at the VON show in Santa Clara, I thought I would submit a highlights message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. Was there any aggressive pricing given for nationwide voip LD? I just lost an Internet customer today who has 6 voice and 2 fax business lines. He is moving to McLeod (regional bankrupt CLEC) for both voice and data. They are putting in a T1 and giving him 2.2 cents a minute for nationwide LD. He had to sign a 3 year contract. The CLEC battle is heating up here. I can't compete when I have to pay more than that for VOIP LD calls that terminate on POTS. Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Scott Laird wrote: On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote: SRTP protects RTP/UDP media with encryption. There are concerns that sending positioning within SIP/UDP will reveal private detailes, like position. Hence the encryption requirement. The position data needs to be given by the ISP in DHCP configuration. This brings up two more questions: 1. What does 'positioning' mean in a SIP context--Google isn't helpful. Is this basically just physical location? If I understand Brian correctly, it will be a global system that can look up the closes 911 service - any where. Possibly latitude and longitude. Drafts out there somewhere, RFCs on it's way before new year. 2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients out there? SIPfoundry got one, another one on SourceForge - maybe they're the same. More information about SRTP and pointers: http://www.voip-info.org/tiki-index.php?page=srtp /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Mark Messmore, Technical Support, University Telcom Inc. wrote: K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade WiSIP, IIRC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Scott Laird wrote: 2. Is anyone working on SRTP for Asterisk? Are there any SRTP clients out there? Checked again, the vovida.org and the sourceforge one are the same. And here's the good news: THey're using a BSD license. That means we can incorporate this library into Asterisk without a licensing problem. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Was there any aggressive pricing given for nationwide voip LD? Level3 had several products, one they called Enhanced which was supposed to also include E911 service. They quoted me about $.01 per minute inbound or outbound nation wide. They said they support the top 300 cities in the US and, of course, have plans to serve every rate center in the US. I also went and talked with ITXC, but the rather bad sales person said they were only really interested in international calling and not domestic LD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spring VON Wrap Up
Bob Klepfer wrote: Mark Messmore, Technical Support, University Telcom Inc. wrote: K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade WiSIP, IIRC After a peek under the hood, I would guess we could have these manufactured over seas for around $1000 USD per unit. It would not be the same to modify the design in any way. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
