Two posts in one day. Wow.
Just as the title says and previous posts have hinted, FreeSWITCH has been added to AstLinux. Please see the official announcement on the AstLinux homepage.
Tuesday, February 17, 2009
AstLinux Updates
Just a few quick AstLinux updates:
1) FreeSWITCH support is just about done and has been committed to trunk. Still needs some testing but things look very promising. The installed binaries are much smaller than I thought (~3MB or so stripped and linked against uClibc). This includes some cool stuff like mod_xml_curl, mod_lua, mod_vmd, mod_snom, and more. As Tony pointed out 1.2MB of that 3MB is mod_sofia! As frustrating as that is at least mod_sofia includes IPV6, UDP, TCP, TLS, session timers, and just about every other relevant SIP standard... ;)
Small note but worth mentioning - FreeSWITCH is the first package to be committed without support for a keydisk of any kind. You need to use unionfs with this one!
2) OpenSER support has been moved to OpenSIPS. I'm still not sure about this one... I've got an existing relationship with Bogdan and all of my projects have been moved to OpenSIPS (instead of Kamailio or SER). We'll see what the community wants on this one (if AstLinux users care about a high performance SIP proxy at all).
3) RTPProxy upddate to 1.1. I love this software.
That's it for now!
1) FreeSWITCH support is just about done and has been committed to trunk. Still needs some testing but things look very promising. The installed binaries are much smaller than I thought (~3MB or so stripped and linked against uClibc). This includes some cool stuff like mod_xml_curl, mod_lua, mod_vmd, mod_snom, and more. As Tony pointed out 1.2MB of that 3MB is mod_sofia! As frustrating as that is at least mod_sofia includes IPV6, UDP, TCP, TLS, session timers, and just about every other relevant SIP standard... ;)
Small note but worth mentioning - FreeSWITCH is the first package to be committed without support for a keydisk of any kind. You need to use unionfs with this one!
2) OpenSER support has been moved to OpenSIPS. I'm still not sure about this one... I've got an existing relationship with Bogdan and all of my projects have been moved to OpenSIPS (instead of Kamailio or SER). We'll see what the community wants on this one (if AstLinux users care about a high performance SIP proxy at all).
3) RTPProxy upddate to 1.1. I love this software.
That's it for now!
Friday, February 13, 2009
The Joys of the Internet
The members of the Yahoo! Financial forum are now referring to me as "little hitler" (it's because I have a German last name, how clever) and suggesting I be sued for libel... That would get interesting. Perhaps they have never heard of The Streisand Effect?
It is unfortunate that what should be a mature, productive discussion has come to this. What is it about the internet that brings out the worst in people?
Anyways, it is very clear this issue should have never left a technical forum. I don't give stock tips; they shouldn't argue RFCs.
It is unfortunate that what should be a mature, productive discussion has come to this. What is it about the internet that brings out the worst in people?
Anyways, it is very clear this issue should have never left a technical forum. I don't give stock tips; they shouldn't argue RFCs.
Yikes! (More on Sonus)
It seems that my Sonus rant has crossed over into some other arenas...
Yahoo! Financial Discussion
I've signed up for an account (kristian.kielhofner -at- yahoo.com) to participate in this thread. Should be interesting.
Yahoo! Financial Discussion
I've signed up for an account (kristian.kielhofner -at- yahoo.com) to participate in this thread. Should be interesting.
Monday, February 9, 2009
More on FreeSWITCH
I've hinted at it for some time:
I've been playing with FreeSWITCH.
Anyone who is reading this should already know what FreeSWITCH is and why someone (such as myself) would be so interested in it. I'm not going to go over all of that again; there are plenty of rave reviews all over the internet. I don't need to write another one (although I probably will some day).
Here's an update on what I've done so far:
1) FreeSWITCH support in AstLinux. Still coming along but much progress has already been made. It compiles cleanly (one more hack for sqlite) and appears to work. More testing soon but I was pleasantly surprised - the build system seems to be just as well designed as the rest of the project. They've done a great job!
2) I hate transcoding. Long, long ago I led an effort to re-record and convert all of the Asterisk prompts to various native file formats to avoid transcoding. More than two years later I'm doing it again for FreeSWITCH although this time I don't have to pay to re-record them all! Luckily they are made available in various sample rates already. I just had to update the script and do the converting. Big thanks to sox, Asterisk/res_convert and FreeSWITCH/Mod_native_file.
3) This one is barely worth mentioning but I've started (SVN branch, that's about it) working to re-implement recqual using FreeSWITCH to place calls. I've got some big plans for this. Let's see how much time I actually have to work on it. Don't expect much progress anytime soon.
4) Various production and consulting projects. Obviously.
As always, expect more to come!
I've been playing with FreeSWITCH.
Anyone who is reading this should already know what FreeSWITCH is and why someone (such as myself) would be so interested in it. I'm not going to go over all of that again; there are plenty of rave reviews all over the internet. I don't need to write another one (although I probably will some day).
Here's an update on what I've done so far:
1) FreeSWITCH support in AstLinux. Still coming along but much progress has already been made. It compiles cleanly (one more hack for sqlite) and appears to work. More testing soon but I was pleasantly surprised - the build system seems to be just as well designed as the rest of the project. They've done a great job!
2) I hate transcoding. Long, long ago I led an effort to re-record and convert all of the Asterisk prompts to various native file formats to avoid transcoding. More than two years later I'm doing it again for FreeSWITCH although this time I don't have to pay to re-record them all! Luckily they are made available in various sample rates already. I just had to update the script and do the converting. Big thanks to sox, Asterisk/res_convert and FreeSWITCH/Mod_native_file.
3) This one is barely worth mentioning but I've started (SVN branch, that's about it) working to re-implement recqual using FreeSWITCH to place calls. I've got some big plans for this. Let's see how much time I actually have to work on it. Don't expect much progress anytime soon.
4) Various production and consulting projects. Obviously.
As always, expect more to come!
Thursday, February 5, 2009
The update you've been waiting for...
UPDATE: Any updates for this and other SIP/RTP issues can be found here.
In my last post over one month ago, I ranted on and on (big surprise, right) about some issues with Sonus equipment we were experiencing. After learning more I should elaborate on "Sonus equipment".
Like many other manufacturers Sonus has multiple products. We'll be talking about their NBS SBC. Many providers use the NBS SBC in conjunction with GSX gateways and PSX route servers. I have no comments about GSX gateways or PSX route servers; this equipment is largely transparent to us "end users". My gripes are with the NBS SBC.
Providers that use Sonus NBS:
- Level(3) (w/ GSX)
- XO (w/ PSX & GSX)
- Global Crossing
- Broadvox
- Many others
If you are using these carriers for SIP services, be aware.
Last time I was talking about timestamps. This time it's far more insidious...
Apparently (as relayed to me from Level(3) engineers) Sonus has a DSP buffer limitation for RTP packet handling. If there is ever more than a 100ms (my experience has shown it to be much less) gap in RTP Sonus will in technical terms, "freak out".
We have now identified four RTP interop issues with Sonus equipment:
1) Sonus requires all RTP packets (events or voice) to have unique timestamps. The RFCs specifically state that not only is it valid to use the same timestamp for various RTP packets, it is ideal in some cases (like events, for example).
2) The RFC 2833 events generated by Sonus equipment are goofy, to put it lightly. The event duration increments do not match the packetization of the voice stream as stated in RFC 2833 and elaborated on in RFC 4733. Specifically, Sonus equipment increments RFC 2833 duration 80 samples
at a time as if the voice stream is 10ms (regardless of what it actually is). I don't know of any other implementations that do this. Even when the audio stream is *clearly* 20 ms (in the SDP, too) Sonus will continue to increment 80 samples at a time.
3) The most recent (and biggest problem) has been caused by the Sonus (seemingly arbitrary) requirement that there never be greater than 100ms gaps in RTP. This is inherently broken behavior for robustness in IP networks.
4) Sonus has yet another issue with RTP timing and sequencing... If a call is brought up with an endpoint that clocks it's own RTP stream (IVR server, for example) everything will be fine. Until the IVR server (or whatever) bridges that channel to another device that also clocks its own RTP. Sonus (probably related to #3 above) will lose sync and drop audio for up to several seconds while it catches up to the new RTP stream. This requires those of us that work with Sonus equipment to rewrite all timestamps and sequence numbers on our equipment; which has the adverse effect of less than optimal jitter buffering (which should ideally be done at each far endpoint).
Asterisk is largely ok with all of these issues, believe it or not. The one that still causes problems is #3. If you are using Asterisk and Sonus gateways, make DAMN SURE that you are using Packet2Packet bridging and that your devices (whatever they may be) implement RFC 2833 the Sonus way. If not...
NO DTMF FOR YOU!
If you are not using Packet2Packet bridging and your events need to traverse the Asterisk core (for features, fixup, or anything else) there will be a variable length RTP gap that often exceeds the Sonus DSP buffer requirement. With gaps in RTP...
NO DTMF FOR YOU!
FreeSWITCH is also ok as long as you avoid #4. FreeSWITCH provides the configuration option to rewrite timestamps and break jitter buffering. If you are using Sonus gateways you should enable it, otherwise...
NO DTMF FOR YOU!
All of this makes me wish I was around back in the old days when there was one telco and all DTMF was inband!
In my last post over one month ago, I ranted on and on (big surprise, right) about some issues with Sonus equipment we were experiencing. After learning more I should elaborate on "Sonus equipment".
Like many other manufacturers Sonus has multiple products. We'll be talking about their NBS SBC. Many providers use the NBS SBC in conjunction with GSX gateways and PSX route servers. I have no comments about GSX gateways or PSX route servers; this equipment is largely transparent to us "end users". My gripes are with the NBS SBC.
Providers that use Sonus NBS:
- Level(3) (w/ GSX)
- XO (w/ PSX & GSX)
- Global Crossing
- Broadvox
- Many others
If you are using these carriers for SIP services, be aware.
Last time I was talking about timestamps. This time it's far more insidious...
Apparently (as relayed to me from Level(3) engineers) Sonus has a DSP buffer limitation for RTP packet handling. If there is ever more than a 100ms (my experience has shown it to be much less) gap in RTP Sonus will in technical terms, "freak out".
We have now identified four RTP interop issues with Sonus equipment:
1) Sonus requires all RTP packets (events or voice) to have unique timestamps. The RFCs specifically state that not only is it valid to use the same timestamp for various RTP packets, it is ideal in some cases (like events, for example).
2) The RFC 2833 events generated by Sonus equipment are goofy, to put it lightly. The event duration increments do not match the packetization of the voice stream as stated in RFC 2833 and elaborated on in RFC 4733. Specifically, Sonus equipment increments RFC 2833 duration 80 samples
at a time as if the voice stream is 10ms (regardless of what it actually is). I don't know of any other implementations that do this. Even when the audio stream is *clearly* 20 ms (in the SDP, too) Sonus will continue to increment 80 samples at a time.
3) The most recent (and biggest problem) has been caused by the Sonus (seemingly arbitrary) requirement that there never be greater than 100ms gaps in RTP. This is inherently broken behavior for robustness in IP networks.
4) Sonus has yet another issue with RTP timing and sequencing... If a call is brought up with an endpoint that clocks it's own RTP stream (IVR server, for example) everything will be fine. Until the IVR server (or whatever) bridges that channel to another device that also clocks its own RTP. Sonus (probably related to #3 above) will lose sync and drop audio for up to several seconds while it catches up to the new RTP stream. This requires those of us that work with Sonus equipment to rewrite all timestamps and sequence numbers on our equipment; which has the adverse effect of less than optimal jitter buffering (which should ideally be done at each far endpoint).
Asterisk is largely ok with all of these issues, believe it or not. The one that still causes problems is #3. If you are using Asterisk and Sonus gateways, make DAMN SURE that you are using Packet2Packet bridging and that your devices (whatever they may be) implement RFC 2833 the Sonus way. If not...
NO DTMF FOR YOU!
If you are not using Packet2Packet bridging and your events need to traverse the Asterisk core (for features, fixup, or anything else) there will be a variable length RTP gap that often exceeds the Sonus DSP buffer requirement. With gaps in RTP...
NO DTMF FOR YOU!
FreeSWITCH is also ok as long as you avoid #4. FreeSWITCH provides the configuration option to rewrite timestamps and break jitter buffering. If you are using Sonus gateways you should enable it, otherwise...
NO DTMF FOR YOU!
All of this makes me wish I was around back in the old days when there was one telco and all DTMF was inband!
Wednesday, January 7, 2009
Heads up!
UPDATE: Any developments on this and other SIP/RTP issues can be found here.
Some serious issues for all of those of you in SIP land:
There is a pretty serious RTP problem with Sonus equipment that has been making the rounds...
Simply put, Sonus equipment will not accept two RTP packets with the same timestamp, even if the sequence number has been properly incremented. According to various RFCs (namely 1889 and 2833) this is perfectly valid and in some cases (like video) desired.
A few slight problems... Many implementations (including Asterisk AND FreeSWITCH) will (did -more on this later) send out RFC 2833 DTMF events with the same timestamp as the last voice RTP packet. This is perfectly valid according to the RFCs mentioned above.
It appears (after my own testing) that Sonus will actually drop BOTH the voice RTP packet and the event packet. After some testing against Sonus gear it was pretty clear that no audio was being passed as long as the DTMF event occured. This makes sense because per RFC2833 a variable length DTMF event must use the same timestamp, increment the sequence counter and increase the duration when it is resent - DO NOT change the timestamp. Oh Sonus.
Both Asterisk and FreeSWITCH have incremented workarounds to address this. They are similar but there is one key difference. Asterisk now (as of SVN 12/15/2008 or so) will always use a unique timestamp for every RTP packet. I guess that solves that problem. FreeSWITCH is slightly smarter about it (as of SVN about the same time, interestingly enough) but I"m worried...
FreeSWITCH will parse the SDP to find the originator line (o=). If it is equal to "Sonus_UAC" FreeSWITCH activates a specific workaround to always send RTP packets with different timestamps. This seems more elegant but I am worried they will have to expand this hack for other equipment in the future (requiring a code change and recompile).
One could argue that Sonus has gotten this far with their current implementation and expected behavior. While it is valid (per the RFCs) to use the same timestamp, it is more /compatible/ to always use different timestamps. That appears to be what most equipment does.
This issue is what (apparantly) caused so many issues for Teliax a while back while they switched from Asterisk to FreeSWITCH. At least that's what I heard. What doesn't make any sense is that Asterisk had the same behavior as FreeSWITCH - they both sent voice and event RTP packets with identical timestamps. So that part doesn't make any sense.
Also, one would like to think that when you provide voice services (which are pretty important to your customers) you would *test* something like DTMF when you were completely switching platforms. I discovered these issues while testing Star2Star with Level(3), for example. I'm glad I was paying attention. Our customers would have been upset with broken DTMF while we updated all of our Asterisk machines (several hundred).
I'm suprised no one noticed this until mid-December or so. It will be interesting to see what other things pop out of this mess...
Some serious issues for all of those of you in SIP land:
There is a pretty serious RTP problem with Sonus equipment that has been making the rounds...
Simply put, Sonus equipment will not accept two RTP packets with the same timestamp, even if the sequence number has been properly incremented. According to various RFCs (namely 1889 and 2833) this is perfectly valid and in some cases (like video) desired.
A few slight problems... Many implementations (including Asterisk AND FreeSWITCH) will (did -more on this later) send out RFC 2833 DTMF events with the same timestamp as the last voice RTP packet. This is perfectly valid according to the RFCs mentioned above.
It appears (after my own testing) that Sonus will actually drop BOTH the voice RTP packet and the event packet. After some testing against Sonus gear it was pretty clear that no audio was being passed as long as the DTMF event occured. This makes sense because per RFC2833 a variable length DTMF event must use the same timestamp, increment the sequence counter and increase the duration when it is resent - DO NOT change the timestamp. Oh Sonus.
Both Asterisk and FreeSWITCH have incremented workarounds to address this. They are similar but there is one key difference. Asterisk now (as of SVN 12/15/2008 or so) will always use a unique timestamp for every RTP packet. I guess that solves that problem. FreeSWITCH is slightly smarter about it (as of SVN about the same time, interestingly enough) but I"m worried...
FreeSWITCH will parse the SDP to find the originator line (o=). If it is equal to "Sonus_UAC" FreeSWITCH activates a specific workaround to always send RTP packets with different timestamps. This seems more elegant but I am worried they will have to expand this hack for other equipment in the future (requiring a code change and recompile).
One could argue that Sonus has gotten this far with their current implementation and expected behavior. While it is valid (per the RFCs) to use the same timestamp, it is more /compatible/ to always use different timestamps. That appears to be what most equipment does.
This issue is what (apparantly) caused so many issues for Teliax a while back while they switched from Asterisk to FreeSWITCH. At least that's what I heard. What doesn't make any sense is that Asterisk had the same behavior as FreeSWITCH - they both sent voice and event RTP packets with identical timestamps. So that part doesn't make any sense.
Also, one would like to think that when you provide voice services (which are pretty important to your customers) you would *test* something like DTMF when you were completely switching platforms. I discovered these issues while testing Star2Star with Level(3), for example. I'm glad I was paying attention. Our customers would have been upset with broken DTMF while we updated all of our Asterisk machines (several hundred).
I'm suprised no one noticed this until mid-December or so. It will be interesting to see what other things pop out of this mess...
Monday, December 22, 2008
Introducing Recqual
I've been waiting to talk about this one for a while.
Several months ago Star2Star was having problems with one of our upstream SIP carriers. We were starting to notice a large increase in the number of one way audio calls our customers were reporting.
When most people think of one way calls their first reaction is to blame SIP. Must be NAT! Must be a firewall! SIP sucks! Etc, etc.
I knew that wasn't the case. I just had to prove it.
I was convinced the problem wasn't SIP/UDP/IP related at all. We had multiple pcaps where we were sending RTP to the appropriate gateway. It just wasn't getting to the PSTN. Where was it going? When was this happening? Which gateways (out of hundreds) were the most problematic? We needed to know and we needed to know quickly.
I came up with and "wrote" recqual over a couple of days. After a few runs we were noticing patterns with problematic RTP endpoint IP addresses. Long story short, once these were identified we worked with the carrier to replace various bits of equipment (DSPs, line cards, etc). The one way audio problem has largely disappeared and we continue to run recqual. If this starts happening again we should know /BEFORE/ our customers do.
Of course I'm using Asterisk to place the calls. The best part of using Asterisk is it's multi-protocol flexibility. You should be able to test just about any combination of voice technologies - G.279a, G711, GSM, SIP, IAX, PRI, FXO, FXO, gtalk/jabber/jingle, skype, etc. The possibilities boggle the mind.
I've just been too busy to get it together and release this to the community - until now.
Tarball with instructions here.
Questions? Comments? Suggestions? Drop me a line.
Several months ago Star2Star was having problems with one of our upstream SIP carriers. We were starting to notice a large increase in the number of one way audio calls our customers were reporting.
When most people think of one way calls their first reaction is to blame SIP. Must be NAT! Must be a firewall! SIP sucks! Etc, etc.
I knew that wasn't the case. I just had to prove it.
I was convinced the problem wasn't SIP/UDP/IP related at all. We had multiple pcaps where we were sending RTP to the appropriate gateway. It just wasn't getting to the PSTN. Where was it going? When was this happening? Which gateways (out of hundreds) were the most problematic? We needed to know and we needed to know quickly.
I came up with and "wrote" recqual over a couple of days. After a few runs we were noticing patterns with problematic RTP endpoint IP addresses. Long story short, once these were identified we worked with the carrier to replace various bits of equipment (DSPs, line cards, etc). The one way audio problem has largely disappeared and we continue to run recqual. If this starts happening again we should know /BEFORE/ our customers do.
Of course I'm using Asterisk to place the calls. The best part of using Asterisk is it's multi-protocol flexibility. You should be able to test just about any combination of voice technologies - G.279a, G711, GSM, SIP, IAX, PRI, FXO, FXO, gtalk/jabber/jingle, skype, etc. The possibilities boggle the mind.
I've just been too busy to get it together and release this to the community - until now.
Tarball with instructions here.
Questions? Comments? Suggestions? Drop me a line.
Sunday, December 21, 2008
Consulting Time Available
I haven't blogged in a while but there is some good news...
I have made some time available for consulting work!
You might not be as excited about it as I am but this is a good thing. I'm looking for interesting projects, people, and companies to work with.
If you or anyone you know might be interested please contact me. Resume, references, etc available on request.
I'll be offering bonus time, discounts, and a few other potential incentives to anyone that lets me blog about my projects and/or release any work under a liberal (read: FOSS) license.
Between my change in schedule and some (hopefully) fun new projects you can all expect to see much more frequent blogging soon!
I have made some time available for consulting work!
You might not be as excited about it as I am but this is a good thing. I'm looking for interesting projects, people, and companies to work with.
If you or anyone you know might be interested please contact me. Resume, references, etc available on request.
I'll be offering bonus time, discounts, and a few other potential incentives to anyone that lets me blog about my projects and/or release any work under a liberal (read: FOSS) license.
Between my change in schedule and some (hopefully) fun new projects you can all expect to see much more frequent blogging soon!
Monday, November 17, 2008
SBCs are Killing SIP
Wow... Over a month since my last post! My how time flies.
No time to reminisce or catch up. I've got a rant that needs to get out - NOW.
SBCs (Session Border Controllers) are killing SIP. Breaking SIP. Smothering SIP. Especially when used by "carriers". Carriers and their SBCs I tells ya.
SBCs, technically, are pretty cool devices. While I certainly understand their purpose they tend to be overused, misconfigured, and misunderstood. Many entities deploy SBCs without any idea of the other components (I'm looking at you, proxies) that make up a well designed SIP network.
Why do I hate SBCs so much?
1) SIP is cool because it is end to end and designed with intelligent endpoints in mind (endpoints that can think for themselves).
2) SIP is very flexible, especially with regards to handling media.
3) Ubiquity.
SBCs (especially when misconfigured) break many of these features:
1) SBCs (by design) hide endpoints from one another. Both endpoints support G.722? The SBC doesn't and it's going to rewrite the SDP with it's capabilities. Too bad.
2) SBCs (by design) handle media. While this can be good often times it isn't and there are other, less drastic ways to ensure quality of media.
3) When the only tool you have is a hammer, every problem starts to look like a nail.
My biggest concerns with SBCs relate to the last point. I swear, there are many providers, enterprises, etc that have deployed SIP in some capacity using ONLY SBCs and simple UACs and UASs. They've never heard of a proxy. Or a registrar. Heck, I'd even go for a signalling-only B2BUA and call it a compromise. Chances are they've never heard of that either.
I have dealt with several devices that break down, utterly fall apart when used with a proxy. I've covered it on this blog before. I'm just too mad to look up the link now. Again, $MANUFACTURER designs and markets a SIP device. They only test it against SBCs and they've (apparently) never heard of a proxy. Guess what happens...
Some poor soul like myself tries to deploy said device in what I consider to be a well designed SIP network. Unfortunately for me, this call path might not involve an SBC. Guess what happens? The device doesn't understand traversing proxies (Record-Route, Via, etc) and does something silly like parse the Contact header when trying to send a response. Call failure and all kinds of brokenness ensue.
So... I talk to $MANUFACTURER and get the standard "We've deployed this device thousands of times and never seen this problem before". Let's assume that's true. I don't know what's more depressing: the fact that they skipped over multiple sections of a basic SIP RFC like 3261 or the fact that no one noticed it for this long because (apparantly) no one uses proxies anymore. Ugh. Gross.
It's not just device manufacturers. Carriers do this too. Often times the actual issue lies with their SBC. Many carriers (especially those using ACME SBCs, it seems) parse To: instead of the Request-URI. Probably because their customers are using SBCs too and Request-URI and To: match. Not so with a proxy. I don't blame the carrier's use of an SBC. This makes sense. That's what they were designed for. However, please test your device and configuration against something other than another SBC.
What happens if your Request-URI and To: don't match? They send a 404! Yet another RFC3261 violation. Section 8.2.2.1 allows for a UAS to route based off To (although it doesn't sound preffered). However, for the love of God, if you are going to deny a request because of the content of a To header, please send a 403 as specified in the RFC. Your 404s are confusing and ignorant. Was it really not found, or are you just routing based off To instead of the Request-URI? Once again I blame SBCs and a world where it's becoming common for SBCs to talk to each other (and nothing else).
This is yet another situation where assumptions are made based on the behavior of SBCs. It's bad. Please stop.
No time to reminisce or catch up. I've got a rant that needs to get out - NOW.
SBCs (Session Border Controllers) are killing SIP. Breaking SIP. Smothering SIP. Especially when used by "carriers". Carriers and their SBCs I tells ya.
SBCs, technically, are pretty cool devices. While I certainly understand their purpose they tend to be overused, misconfigured, and misunderstood. Many entities deploy SBCs without any idea of the other components (I'm looking at you, proxies) that make up a well designed SIP network.
Why do I hate SBCs so much?
1) SIP is cool because it is end to end and designed with intelligent endpoints in mind (endpoints that can think for themselves).
2) SIP is very flexible, especially with regards to handling media.
3) Ubiquity.
SBCs (especially when misconfigured) break many of these features:
1) SBCs (by design) hide endpoints from one another. Both endpoints support G.722? The SBC doesn't and it's going to rewrite the SDP with it's capabilities. Too bad.
2) SBCs (by design) handle media. While this can be good often times it isn't and there are other, less drastic ways to ensure quality of media.
3) When the only tool you have is a hammer, every problem starts to look like a nail.
My biggest concerns with SBCs relate to the last point. I swear, there are many providers, enterprises, etc that have deployed SIP in some capacity using ONLY SBCs and simple UACs and UASs. They've never heard of a proxy. Or a registrar. Heck, I'd even go for a signalling-only B2BUA and call it a compromise. Chances are they've never heard of that either.
I have dealt with several devices that break down, utterly fall apart when used with a proxy. I've covered it on this blog before. I'm just too mad to look up the link now. Again, $MANUFACTURER designs and markets a SIP device. They only test it against SBCs and they've (apparently) never heard of a proxy. Guess what happens...
Some poor soul like myself tries to deploy said device in what I consider to be a well designed SIP network. Unfortunately for me, this call path might not involve an SBC. Guess what happens? The device doesn't understand traversing proxies (Record-Route, Via, etc) and does something silly like parse the Contact header when trying to send a response. Call failure and all kinds of brokenness ensue.
So... I talk to $MANUFACTURER and get the standard "We've deployed this device thousands of times and never seen this problem before". Let's assume that's true. I don't know what's more depressing: the fact that they skipped over multiple sections of a basic SIP RFC like 3261 or the fact that no one noticed it for this long because (apparantly) no one uses proxies anymore. Ugh. Gross.
It's not just device manufacturers. Carriers do this too. Often times the actual issue lies with their SBC. Many carriers (especially those using ACME SBCs, it seems) parse To: instead of the Request-URI. Probably because their customers are using SBCs too and Request-URI and To: match. Not so with a proxy. I don't blame the carrier's use of an SBC. This makes sense. That's what they were designed for. However, please test your device and configuration against something other than another SBC.
What happens if your Request-URI and To: don't match? They send a 404! Yet another RFC3261 violation. Section 8.2.2.1 allows for a UAS to route based off To (although it doesn't sound preffered). However, for the love of God, if you are going to deny a request because of the content of a To header, please send a 403 as specified in the RFC. Your 404s are confusing and ignorant. Was it really not found, or are you just routing based off To instead of the Request-URI? Once again I blame SBCs and a world where it's becoming common for SBCs to talk to each other (and nothing else).
This is yet another situation where assumptions are made based on the behavior of SBCs. It's bad. Please stop.
Thursday, October 9, 2008
Submit Your SIP
Ever since I've started blogging and talking about SIP people have come out of the woodwork with SIP interop problems.
After giving a talk about SIP at Astricon 2008 I received several e-mails from audience members with specific SIP issues. I LOVE getting these e-mails.
Why? I love working on SIP issues. With all of the devices using SIP there is no shortage of interop problems. Just today a guy on the Asterisk mailing list had a problem with his Cisco AS5300 and Asterisk 1.2 Usually that wouldn't be a problem at all - many people (including myself) use this combination of hardware with great success.
Why was he having problems? His AS5300 was configured for GTD and Asterisk 1.2 (apparently) doesn't handle multipart SIP bodies very well. I was able to find a patch to Asterisk 1.4 to improve multipart body parsing. That was a fun one.
I got to thinking... There should be a place where people can exchange specific SIP interop tips and notes. Otherwise how are we supposed to get anything to work!?!?
I came up with such a place and it's called SubmitYourSip.com. I' ve started to fill it in a little but hopefully (with time) it will become somewhat of a SIP wiki (with a focus on interop, of course).
I'm just getting started on it but I'll be working on my MediaWiki syntax and going back through my e-mail to dig out some of these examples.
After giving a talk about SIP at Astricon 2008 I received several e-mails from audience members with specific SIP issues. I LOVE getting these e-mails.
Why? I love working on SIP issues. With all of the devices using SIP there is no shortage of interop problems. Just today a guy on the Asterisk mailing list had a problem with his Cisco AS5300 and Asterisk 1.2 Usually that wouldn't be a problem at all - many people (including myself) use this combination of hardware with great success.
Why was he having problems? His AS5300 was configured for GTD and Asterisk 1.2 (apparently) doesn't handle multipart SIP bodies very well. I was able to find a patch to Asterisk 1.4 to improve multipart body parsing. That was a fun one.
I got to thinking... There should be a place where people can exchange specific SIP interop tips and notes. Otherwise how are we supposed to get anything to work!?!?
I came up with such a place and it's called SubmitYourSip.com. I' ve started to fill it in a little but hopefully (with time) it will become somewhat of a SIP wiki (with a focus on interop, of course).
I'm just getting started on it but I'll be working on my MediaWiki syntax and going back through my e-mail to dig out some of these examples.
Friday, September 19, 2008
A preview: performance tests
I'm headed out the door for some sushi but I thought I'd drop in to give you an idea of what I'm working on for my next blog post. I'm hungry so let's keep this short and sweet: receive interrupt mitigation and its effects on Linux media applications.
In general I'm a big fan of receive interrupt mitigation. I'll trade some delay for a substantial decrease in system CPU time spent servicing interrupts resulting in the ability to handle more calls. I didn't just come to this one day, I've done some tests in the past to verify this.
However, I've never done a large scale test on regular, server class hardware. Usually just Asterisk on an embedded system. It usually works out well. This is why, by default, all ethernet adapters that support NAPI in Linux are enabled in AstLinux by default.
The folks at TransNexus spend a fair amount of time testing OpenSER/Kamailio/OpenSips performance. Today Jim Dalton posted the results of another test to the Kamailio User's mailing list. I replied to his post with so many questions I figured it might be time for me to lab this up myself and test my theories about interrupt handling (in Linux, specifically).
If those brighly colored rolls of fish weren't so distracting and delicious I'd promise to think about all of this over dinner. Unfortunately it will have to wait until tomorrow...
In general I'm a big fan of receive interrupt mitigation. I'll trade some delay for a substantial decrease in system CPU time spent servicing interrupts resulting in the ability to handle more calls. I didn't just come to this one day, I've done some tests in the past to verify this.
However, I've never done a large scale test on regular, server class hardware. Usually just Asterisk on an embedded system. It usually works out well. This is why, by default, all ethernet adapters that support NAPI in Linux are enabled in AstLinux by default.
The folks at TransNexus spend a fair amount of time testing OpenSER/Kamailio/OpenSips performance. Today Jim Dalton posted the results of another test to the Kamailio User's mailing list. I replied to his post with so many questions I figured it might be time for me to lab this up myself and test my theories about interrupt handling (in Linux, specifically).
If those brighly colored rolls of fish weren't so distracting and delicious I'd promise to think about all of this over dinner. Unfortunately it will have to wait until tomorrow...
Friday, September 12, 2008
Distro Wars!
I don't like to get into Distro Wars... Nothing is more pathetic than a bunch of FOSS geeks sitting around getting religious about:
- Distros
- Editors
- Star Wars
Ok, ok I started to embrace the stereotype a little towards the end there but if you've ever seen one of these epic battles with your own eyes you just might relate it to Star Wars too.
There are always far more serious things going on in the world and these software hippies sit around and argue about what software to edit ASCII with. Ridiculous.
However, even I will throw down when people bring up the worst idea for a distro ever:
Fedora
There, I said it. I've officially fired my first shot in a war that has been raging in the Linux community since, oh, 1991 or so.
What's wrong with Fedora, you ask? Of all of the other hundreds of distros, why would I single out Fedora and waste my time writing about it? I'll tell you why:
Much like the iPhone, it's a joke.
I CRINGE. ABSOULTELY CRINGE when I see someone trying to do something serious with Fedora. At this very minute (3am), I am typing a blog post when I should be sleeping. Why you ask? Because I just saw a post on Asterisk-Users with yet another poor soul trying to bring up Asterisk on a Fedora system (Fedora 9). I couldn't possibly sleep knowing the abomination that is Fedora+Asterisk continues.
This is a perfect example to illustrate why Fedora is such a joke. One of the fundamental principles of Linux is reliability. One of the fundamental principles of telephony is reliability. By installing Asterisk on Fedora you are flying in the face of 18 years of Linux and 100+ years of telephony.
Fedora is BLEEDING EDGE. It serves as a test bed for RedHat's next REAL Linux release. How do you feel being a tester for what is sold as a commercial product by a profitable company? Bugs in Fedora are found and fixed quickly...
For six months
You can install Fedora and deal with the usual issues in beta software. Once things finally settle down (after six months or so), you get to upgrade to the next release and start all over again!
Sounds like a great plan for a server or PBX, right?
Hosting companies sell packages based on Fedora. Shame on them. People install PBX systems with Fedora. Double shame on them. The average life span for a PBX is seven years. This means that your Fedora Asterisk system is supported with updates for %7 of the typical PBX lifespan.
CentOS/RHEL/Ubuntu LTS and several other Novell/etc offerings are supported for five years or more.
Granted Fedora has it's place. It makes a nice toy, much like the iPhone. If you want to play with cutting edge Linux, Fedora is for you and it might work on a test system, desktop, laptop, etc.
But please. Please. PLEASE do not install it on a server and don't even think of using it for Asterisk. ANYTHING else will do. Seriously, I don't have a problem with any other distro. Pick one.
- Distros
- Editors
- Star Wars
Ok, ok I started to embrace the stereotype a little towards the end there but if you've ever seen one of these epic battles with your own eyes you just might relate it to Star Wars too.
There are always far more serious things going on in the world and these software hippies sit around and argue about what software to edit ASCII with. Ridiculous.
However, even I will throw down when people bring up the worst idea for a distro ever:
Fedora
There, I said it. I've officially fired my first shot in a war that has been raging in the Linux community since, oh, 1991 or so.
What's wrong with Fedora, you ask? Of all of the other hundreds of distros, why would I single out Fedora and waste my time writing about it? I'll tell you why:
Much like the iPhone, it's a joke.
I CRINGE. ABSOULTELY CRINGE when I see someone trying to do something serious with Fedora. At this very minute (3am), I am typing a blog post when I should be sleeping. Why you ask? Because I just saw a post on Asterisk-Users with yet another poor soul trying to bring up Asterisk on a Fedora system (Fedora 9). I couldn't possibly sleep knowing the abomination that is Fedora+Asterisk continues.
This is a perfect example to illustrate why Fedora is such a joke. One of the fundamental principles of Linux is reliability. One of the fundamental principles of telephony is reliability. By installing Asterisk on Fedora you are flying in the face of 18 years of Linux and 100+ years of telephony.
Fedora is BLEEDING EDGE. It serves as a test bed for RedHat's next REAL Linux release. How do you feel being a tester for what is sold as a commercial product by a profitable company? Bugs in Fedora are found and fixed quickly...
For six months
You can install Fedora and deal with the usual issues in beta software. Once things finally settle down (after six months or so), you get to upgrade to the next release and start all over again!
Sounds like a great plan for a server or PBX, right?
Hosting companies sell packages based on Fedora. Shame on them. People install PBX systems with Fedora. Double shame on them. The average life span for a PBX is seven years. This means that your Fedora Asterisk system is supported with updates for %7 of the typical PBX lifespan.
CentOS/RHEL/Ubuntu LTS and several other Novell/etc offerings are supported for five years or more.
Granted Fedora has it's place. It makes a nice toy, much like the iPhone. If you want to play with cutting edge Linux, Fedora is for you and it might work on a test system, desktop, laptop, etc.
But please. Please. PLEASE do not install it on a server and don't even think of using it for Asterisk. ANYTHING else will do. Seriously, I don't have a problem with any other distro. Pick one.
Monday, August 25, 2008
Free cell phones
Just in case you thought I was selling out to the US wireless industry with my previous post, check this out:
After looking over some phones on the Nokia website, I thought of all of those "free" phones carriers like to give away.
Here's one for comparison. The Nokia 6085 is offered in the US by AT&T wireless. Their website says it retails for $189.99. My gosh! Oh but don't you worry, as long as you sign up for two years of service (and do so online) we'll discount our $190 phone to $39.99 and then give you a $39.99 discount (you're buying online, remember). OMG! Free phone! See how that works?
Funny enough, Nokia offers the phone (just the phone) on their US website for $118. That means that right off the bat, AT&T wireless is jacking the price of the phone up $60 just for the pleasure of buying it from them.
Some of you might say "Hey, a $60 markup isn't that bad". Yeah right. AT&T wireless is NOT paying $118 for that phone. I wonder how many of them they sell and what kind of special pricing Nokia gives them. Probably not even close. Probably not even half that. I bet AT&T still makes money at the $39.99 price. I also wonder how many they sell at $189.99...
Here's the catch. They are going to give you the same contract and charge you the same price whether you get the "free" phone or bring your own. That's what sucks about the wireless industry in the US. Unlike the rest of the world, Americans can't be bothered to buy their own phone and bring it to the carrier for service. Maybe it's because we've got some different wireless standards and there could be confusion (iDEN, CDMA, GSM, etc). More than likely it's because the wireless carriers know they can make out like bandits otherwise.
The coupling of the phone to the service is inherently wrong and evil. Some carriers (Verizon Wireless) don't even activate phones who's ESNs they don't have in a database. All to "protect the network". Yeah right. If your network is going to go down because I want to use a CDMA cell phone I bought from Sprint a year ago you've got some serious network issues...
At least there is some light at the end of the tunnel. Soon enough cell phone carriers will have to pro-rate early termination fees. It's tough to lock someone into a contract with a $200 early termination fee to cover the cost of the $40 phone you "gave away" almost two years ago.
After looking over some phones on the Nokia website, I thought of all of those "free" phones carriers like to give away.
Here's one for comparison. The Nokia 6085 is offered in the US by AT&T wireless. Their website says it retails for $189.99. My gosh! Oh but don't you worry, as long as you sign up for two years of service (and do so online) we'll discount our $190 phone to $39.99 and then give you a $39.99 discount (you're buying online, remember). OMG! Free phone! See how that works?
Funny enough, Nokia offers the phone (just the phone) on their US website for $118. That means that right off the bat, AT&T wireless is jacking the price of the phone up $60 just for the pleasure of buying it from them.
Some of you might say "Hey, a $60 markup isn't that bad". Yeah right. AT&T wireless is NOT paying $118 for that phone. I wonder how many of them they sell and what kind of special pricing Nokia gives them. Probably not even close. Probably not even half that. I bet AT&T still makes money at the $39.99 price. I also wonder how many they sell at $189.99...
Here's the catch. They are going to give you the same contract and charge you the same price whether you get the "free" phone or bring your own. That's what sucks about the wireless industry in the US. Unlike the rest of the world, Americans can't be bothered to buy their own phone and bring it to the carrier for service. Maybe it's because we've got some different wireless standards and there could be confusion (iDEN, CDMA, GSM, etc). More than likely it's because the wireless carriers know they can make out like bandits otherwise.
The coupling of the phone to the service is inherently wrong and evil. Some carriers (Verizon Wireless) don't even activate phones who's ESNs they don't have in a database. All to "protect the network". Yeah right. If your network is going to go down because I want to use a CDMA cell phone I bought from Sprint a year ago you've got some serious network issues...
At least there is some light at the end of the tunnel. Soon enough cell phone carriers will have to pro-rate early termination fees. It's tough to lock someone into a contract with a $200 early termination fee to cover the cost of the $40 phone you "gave away" almost two years ago.
Nokia isn't ditching SIP
Someone sent me a link to a blog post about Nokia "turning it's back on VoIP". I know I can get pretty emotional from time to time, but at least I'm accurate when I do. At least I feel like I am, and that's all that matters, right?
Anyways, let's look at this post. The author points out that the new N78 and N96 no longer include the Symbian SIP client. This must mean Nokia is finally giving in to pressure from cell carriers in some huge scheme to enslave the mobile phone subscriber and direct all talk time over their network.
Follow up comments clarify the situation a bit. Nokia simply removed the interface to the SIP stack on the N series. It's still in the firmware and available to any third party developers, so the most "threatening" apps (Truphone, Gizmo, etc) will continue to work. People that just want to configure the phone to connect to their corporate/"personal" PBX will be out of luck. I love saying "personal" PBX. I have one. So do many of my friends. How geeky is that?
Why did the N series have a SIP interface in the first place? Anyone who has ever read my iPhone review knows that I believe teenagers and hipsters control a large chunk of the cellphone market which breaks down into three parts:
- Free (throw away) phones for moms, grandparents, and kids. Sign up for a two year contract and they're yours to keep! What a deal.
- Flashy phones for teenagers and hipsters (it's a better MP3 player/camera than phone)
- Serious business phones (Blackberry, Windows Mobile, most Symbian, etc)
The N series is (and always has been) a hipster phone. It's more likely to compete with the iPhone than the Blackberry or Nokia E-Series.
Speaking of the E-Series... The new Nokia E-71 still includes the SIP client. Nokia turning it's back on VoIP? I don't think so. Nokia learning a little more about their customers? Much more likely.
Anyways, let's look at this post. The author points out that the new N78 and N96 no longer include the Symbian SIP client. This must mean Nokia is finally giving in to pressure from cell carriers in some huge scheme to enslave the mobile phone subscriber and direct all talk time over their network.
Follow up comments clarify the situation a bit. Nokia simply removed the interface to the SIP stack on the N series. It's still in the firmware and available to any third party developers, so the most "threatening" apps (Truphone, Gizmo, etc) will continue to work. People that just want to configure the phone to connect to their corporate/"personal" PBX will be out of luck. I love saying "personal" PBX. I have one. So do many of my friends. How geeky is that?
Why did the N series have a SIP interface in the first place? Anyone who has ever read my iPhone review knows that I believe teenagers and hipsters control a large chunk of the cellphone market which breaks down into three parts:
- Free (throw away) phones for moms, grandparents, and kids. Sign up for a two year contract and they're yours to keep! What a deal.
- Flashy phones for teenagers and hipsters (it's a better MP3 player/camera than phone)
- Serious business phones (Blackberry, Windows Mobile, most Symbian, etc)
The N series is (and always has been) a hipster phone. It's more likely to compete with the iPhone than the Blackberry or Nokia E-Series.
Speaking of the E-Series... The new Nokia E-71 still includes the SIP client. Nokia turning it's back on VoIP? I don't think so. Nokia learning a little more about their customers? Much more likely.
Thursday, August 7, 2008
OpenSER Update
I just got a very interesting Anonymous comment to my last post. One of my BIG questions was answered:
Kamailio or OpenSIPS?
For me that question was answered - OpenSIPS. Star2Star has purchased a block of consulting time from Voice System for OpenSER support. We used it once to flush out a bug in OpenSER 1.2. Other than that we have it for a little business security. It's nice to have an official avenue for support. Plus we like (financially) supporting Open Source software. It gives us so much, we should give a little back.
Anyways, this last post makes it pretty clear: OpenSIPS is Voice System's product. That's all I needed to hear. Thanks Anonymous commenter/blog reader!
Kamailio or OpenSIPS?
For me that question was answered - OpenSIPS. Star2Star has purchased a block of consulting time from Voice System for OpenSER support. We used it once to flush out a bug in OpenSER 1.2. Other than that we have it for a little business security. It's nice to have an official avenue for support. Plus we like (financially) supporting Open Source software. It gives us so much, we should give a little back.
Anyways, this last post makes it pretty clear: OpenSIPS is Voice System's product. That's all I needed to hear. Thanks Anonymous commenter/blog reader!
Tuesday, August 5, 2008
More OpenSER Drama
My last rant on this blog covered the OpenSER name change to Kamailio. It's pretty obvious how I felt about it and it's even more obvious how upset (for lack of a better word) I was with the selection of the new name.
These new developments make a name change pale in comparison. This time I have something much more serious to fret over:
There has been a fork of OpenSER.
First some history. OpenSER started life as SER. Some time ago OpenSER was forked from SER for good reason (and a common one) - the company "sponsoring" the development of SER didn't understand Open Source. Community input was ignored. Patches took forever to be applied. IPTel just didn't get it. The developers (just about all of them) set out on their own to form OpenSER and create an Open Source friendly company to serve the needs of the OpenSER community - paid support, development, consulting, etc.
Almost everyone I know has (by now) abondoned SER and moved to OpenSER. Which is good because this fork (like any other) created a certain amount of fragmentation in the community:
- documentation
- support (mailing lists, etc)
- thirdy party support
- name recognition
Even with the SER and OpenSER projects generally moving in the same direction it was clear they couldn't be treated the same. Various changes were introduced in the configuration for each piece of software. They were (and are) mostly trivial, but you can't simply move an OpenSER configuration to a SER system (or vice-versa). You also can't ask an OpenSER question on a SER list (obviously). Granted most people are subscribed to both and can provide expertise on either product but it still creates a headache for the user - fragmentation of expertise, documentation, support, etc.
As I've said this doesn't seem to be that much of a problem anymore. OpenSER (Kamailio) is clearly where it's at. It's hard for me to describe how much respect I have for this software and it's developers. It is one of the most impressive products I have ever come across. It does what it is designed to do better than just about anything else I've ever seen.
As I have come to use OpenSER more and understand it better the activist in me begins to emerge. I feel pain anytime I see someone use a product where OpenSER could clearly do the job better. OpenSER doesn't get as much credit or use as it should. I can only theorize why this is but I do know one thing:
Forking doesn't help.
Any traction (that's for you, JJ) OpenSER has made over the last few years is being seriously threatened by these political shenanigans. In the last month or so I've reviewed the first OpenSER book and solicted OpenSER contributions for the upcoming O'Reilly Asterisk Cookbook. I was starting to feel like OpenSER was finally headed towards getting the exposure it deserves.
Most developers probably don't care about these things (exposure). They should, and if they don't the project admins should. And if they don't the biz guys at the company sponsoring the development should. Here's why.
Exposure/"traction":
- Attracts more users
-- More users for testing, bug reports, etc
-- More users to write documentation
-- More users to create revenue for the sponsoring company (consulting, etc)
- Attracts more developers. Open Source development is largely ego driven and the larger and more visible the project, the more developers (both good and bad) you attract.
- Attracts more third party interest in the project. Open Source software has a lot of holes for real business use. There is a huge potential for third party projects for OpenSER. Billing systems, desktop call managers, GUIs, etc.
Neither the SER community, OpenSER community, or OpenSIPS community are large enough to sustain this fragmentation. Sure they very well might survive but they won't be what they should.
Now what? There is an OpenSER book, but no OpenSER. There is Kamailio. I hope %100 of the users figure that out. I hope the publisher of the existing OpenSER book figures out how to deal with that. What happens if everyone (well, almost everyone) bails on OpenSER/Kamailio and moves to OpenSIPS? Now anyone providing support for these other OSPs (Open Source Proxies) has to decide which they will support. SER? Kamailio? OpenSIPS? All of them? None of them? That's what worries me.
I haven't decided what I'm going to do. I still use and recommend OpenSER 1.2 so none of this affects me - yet. Of course I'll be watching this closely. I will tell you one thing... I like the name OpenSIPS more than Kamailio...
These new developments make a name change pale in comparison. This time I have something much more serious to fret over:
There has been a fork of OpenSER.
First some history. OpenSER started life as SER. Some time ago OpenSER was forked from SER for good reason (and a common one) - the company "sponsoring" the development of SER didn't understand Open Source. Community input was ignored. Patches took forever to be applied. IPTel just didn't get it. The developers (just about all of them) set out on their own to form OpenSER and create an Open Source friendly company to serve the needs of the OpenSER community - paid support, development, consulting, etc.
Almost everyone I know has (by now) abondoned SER and moved to OpenSER. Which is good because this fork (like any other) created a certain amount of fragmentation in the community:
- documentation
- support (mailing lists, etc)
- thirdy party support
- name recognition
Even with the SER and OpenSER projects generally moving in the same direction it was clear they couldn't be treated the same. Various changes were introduced in the configuration for each piece of software. They were (and are) mostly trivial, but you can't simply move an OpenSER configuration to a SER system (or vice-versa). You also can't ask an OpenSER question on a SER list (obviously). Granted most people are subscribed to both and can provide expertise on either product but it still creates a headache for the user - fragmentation of expertise, documentation, support, etc.
As I've said this doesn't seem to be that much of a problem anymore. OpenSER (Kamailio) is clearly where it's at. It's hard for me to describe how much respect I have for this software and it's developers. It is one of the most impressive products I have ever come across. It does what it is designed to do better than just about anything else I've ever seen.
As I have come to use OpenSER more and understand it better the activist in me begins to emerge. I feel pain anytime I see someone use a product where OpenSER could clearly do the job better. OpenSER doesn't get as much credit or use as it should. I can only theorize why this is but I do know one thing:
Forking doesn't help.
Any traction (that's for you, JJ) OpenSER has made over the last few years is being seriously threatened by these political shenanigans. In the last month or so I've reviewed the first OpenSER book and solicted OpenSER contributions for the upcoming O'Reilly Asterisk Cookbook. I was starting to feel like OpenSER was finally headed towards getting the exposure it deserves.
Most developers probably don't care about these things (exposure). They should, and if they don't the project admins should. And if they don't the biz guys at the company sponsoring the development should. Here's why.
Exposure/"traction":
- Attracts more users
-- More users for testing, bug reports, etc
-- More users to write documentation
-- More users to create revenue for the sponsoring company (consulting, etc)
- Attracts more developers. Open Source development is largely ego driven and the larger and more visible the project, the more developers (both good and bad) you attract.
- Attracts more third party interest in the project. Open Source software has a lot of holes for real business use. There is a huge potential for third party projects for OpenSER. Billing systems, desktop call managers, GUIs, etc.
Neither the SER community, OpenSER community, or OpenSIPS community are large enough to sustain this fragmentation. Sure they very well might survive but they won't be what they should.
Now what? There is an OpenSER book, but no OpenSER. There is Kamailio. I hope %100 of the users figure that out. I hope the publisher of the existing OpenSER book figures out how to deal with that. What happens if everyone (well, almost everyone) bails on OpenSER/Kamailio and moves to OpenSIPS? Now anyone providing support for these other OSPs (Open Source Proxies) has to decide which they will support. SER? Kamailio? OpenSIPS? All of them? None of them? That's what worries me.
I haven't decided what I'm going to do. I still use and recommend OpenSER 1.2 so none of this affects me - yet. Of course I'll be watching this closely. I will tell you one thing... I like the name OpenSIPS more than Kamailio...
Monday, July 28, 2008
OpenSER Name Change
OpenSER has changed their name to Kamailio.
Name changes are tough. This name change is especially tough and dare I even say, personally frustrating.
Working with Star2Star is tough. We have a revolutionary architecture. We operate very differently from most other providers. Heck, we operate differently than most other companies! Don't get me wrong, this is one of the many things I like about us. That's why we're changing what people think about phone (communications) companies.
One of the main drawbacks of these differences (read: advantages) is working with legacy telco. Whether it's legacy telco people, equipment, process, or organization, I have had many, many frustrating clashes with telecom. Frustrations with technology excite me. Frustrations with people drive me crazy. People can't be fixed as easily.
This name change only exacerbates what I see as an already BIG problem. How am I going to talk to veteran telecom executives about a product who's name I can't even pronounce? Even if I do learn how to pronounce it I'll never get past it's awkwardness.
I am very proud of OpenSER. OpenSER has been crucial in making Star2Star what it is. OpenSER is amazing and I have the utmost respect for everyone involved with the project.
I am ashamed of Kamailio. Same project. Same people. Same awesome, quality software. It just doesn't feel the same. Unfortunately this is my emotional reaction to this name change. I can't help how I feel and I'm sure other people (like Alex) feel the same way.
OpenSER is a phenomenal piece of software that commands respect. It's name should too.
Maybe we'll start a new "support" mailing list. Not for people needing technical support for OpenSER/Kamailio, but support for those of us who have been personally affected by this name change.
Wow, now I'm just being dramatic!
Name changes are tough. This name change is especially tough and dare I even say, personally frustrating.
Working with Star2Star is tough. We have a revolutionary architecture. We operate very differently from most other providers. Heck, we operate differently than most other companies! Don't get me wrong, this is one of the many things I like about us. That's why we're changing what people think about phone (communications) companies.
One of the main drawbacks of these differences (read: advantages) is working with legacy telco. Whether it's legacy telco people, equipment, process, or organization, I have had many, many frustrating clashes with telecom. Frustrations with technology excite me. Frustrations with people drive me crazy. People can't be fixed as easily.
This name change only exacerbates what I see as an already BIG problem. How am I going to talk to veteran telecom executives about a product who's name I can't even pronounce? Even if I do learn how to pronounce it I'll never get past it's awkwardness.
I am very proud of OpenSER. OpenSER has been crucial in making Star2Star what it is. OpenSER is amazing and I have the utmost respect for everyone involved with the project.
I am ashamed of Kamailio. Same project. Same people. Same awesome, quality software. It just doesn't feel the same. Unfortunately this is my emotional reaction to this name change. I can't help how I feel and I'm sure other people (like Alex) feel the same way.
OpenSER is a phenomenal piece of software that commands respect. It's name should too.
Maybe we'll start a new "support" mailing list. Not for people needing technical support for OpenSER/Kamailio, but support for those of us who have been personally affected by this name change.
Wow, now I'm just being dramatic!
Wednesday, July 23, 2008
IPv6
I've been playing with IPv6 a bit in the last couple of days and by playing I mean:
- Setup IPv6 tunnel with Hurricane Electric
- Configured Cisco 2811 for Iv6 tunnels (both ends)
- Subnetted (is that a word?) our /48 from HE
- Configured tunnels in Linux with iproute2
- Used radvd in Linux
- Setup AAAAs for services
- Added/enabled IPv6 in AstLinux
- Played with ip6tables
- Worked on my super-secret IPv6 project (more on that later)
Yep, it's been an IPv6 week. As of right now I've got a main IPv6 tunnel from HE (in Dallas) coming into my 2811 in Tampa. I have a /48 routing down that to other tunnels providing multiple /64s to several locations. I'm setup for providing more IPv6 tunnels and /64s in the future from my 2811 (with or without connectivity to the IPv6 net at large).
I've got IPv6 in the datacenter. I've got IPv6 in the office. I've got IPv6 at home. I've got IPv6 everywhere and a TON of IP addresses to boot. It's really pretty cool and other than my funky tunnel configuration (which I actually kind of like) it's pretty easy. Once I've setup the tunnels I just route the appropriate /64s down to each PtP address for each tunnel at each location. It's a bit of a hub and spoke configuration but it works very well so far. Of course it helps when your tunnel gateway (my 2811) has seven upstream IPv4 carriers.
I've also added IPv6 to AstLinux:
- Kernel (IPv6, mobile IPv6, "41" tunnels, netfilter, etc)
- C library (uClibc)
- Busybox (apps in general, ping6, etc)
- mini_httpd
- OpenSSH
- ntpd
- stunnel
- rsync
- php
- libpcap
- tcpdump
- dnsmasq (needs testing)
- nmap
- radvd
- and more
The IPv6 kernel module alias is disabled by default. Anyone that wants to use IPv6 in AstLinux will have to enable it via (you guessed it) rc.conf. It could use some more testing (hint, hint) but so far it looks pretty good.
There was one more thing I was going to talk about... Of yeah, my "super-secret IPv6 project"... I'll have another post for that soon...
- Setup IPv6 tunnel with Hurricane Electric
- Configured Cisco 2811 for Iv6 tunnels (both ends)
- Subnetted (is that a word?) our /48 from HE
- Configured tunnels in Linux with iproute2
- Used radvd in Linux
- Setup AAAAs for services
- Added/enabled IPv6 in AstLinux
- Played with ip6tables
- Worked on my super-secret IPv6 project (more on that later)
Yep, it's been an IPv6 week. As of right now I've got a main IPv6 tunnel from HE (in Dallas) coming into my 2811 in Tampa. I have a /48 routing down that to other tunnels providing multiple /64s to several locations. I'm setup for providing more IPv6 tunnels and /64s in the future from my 2811 (with or without connectivity to the IPv6 net at large).
I've got IPv6 in the datacenter. I've got IPv6 in the office. I've got IPv6 at home. I've got IPv6 everywhere and a TON of IP addresses to boot. It's really pretty cool and other than my funky tunnel configuration (which I actually kind of like) it's pretty easy. Once I've setup the tunnels I just route the appropriate /64s down to each PtP address for each tunnel at each location. It's a bit of a hub and spoke configuration but it works very well so far. Of course it helps when your tunnel gateway (my 2811) has seven upstream IPv4 carriers.
I've also added IPv6 to AstLinux:
- Kernel (IPv6, mobile IPv6, "41" tunnels, netfilter, etc)
- C library (uClibc)
- Busybox (apps in general, ping6, etc)
- mini_httpd
- OpenSSH
- ntpd
- stunnel
- rsync
- php
- libpcap
- tcpdump
- dnsmasq (needs testing)
- nmap
- radvd
- and more
The IPv6 kernel module alias is disabled by default. Anyone that wants to use IPv6 in AstLinux will have to enable it via (you guessed it) rc.conf. It could use some more testing (hint, hint) but so far it looks pretty good.
There was one more thing I was going to talk about... Of yeah, my "super-secret IPv6 project"... I'll have another post for that soon...
Thursday, July 17, 2008
One Week
One week - almost. That's how long it took me to realize that Maddox is right.
This time last week I was plotting how I could sneak into line to get one of those new iPhones. No it's not because I'm some Apply fanboy or some moron that can't do basic math (I'm not paying for the phone or the service). It's because my company is excited about what we might be able to do with the Apps store and the iPhone SDK. Trust me, it's really cool. That's all I can say right now.
I knew we needed at least a couple of these things (iPhones) and I didn't want to get stuck without them. The app we want to do is awesome and I wanted to get to work on it. That meant I was getting an iPhone. I had two choices:
1) Carry two phones
2) Port my number to our AT&T corporate plan and cancel T-Mobile
I wasn't going to carry two phones. A guy I work with carries at least two and it seems like a hassle. I HATE hassle. So it looks like I was going to port my number to AT&T and leave T-Mobile.
I'll skip the next part. We all know what happened here. Quick summary:
- In line at 7:45am
- Out of store by 10:30am (16GB white)
- Phone activated by 3:30pm
Let's just say I couldn't believe the activation problems. I guess Apple and AT&T were too busy marketing the hell out of this thing to remember to buy more servers or upgrade their network. Unbelievable.
Now that I've had the iPhone 3G for almost a week, here's what I've learned. Looking around the web, everyone already knows this:
- The camera sucks. Stills are grainy and often blurry. It can't do video at all. The Motorola RAZR could do video - five years ago. Today any free phone with a service plan can do video.
- Battery life sucks. For the first time in years, I actually worry about my cell phone battery.
- I don't care what anyone says, the keyboard sucks. It does ok considering there are no buttons but Apple needs to learn when they're wrong. They were wrong with the first gen, and they still are. It is not usable for any real work. I almost got to the point where I started sending incomprehensible teenage TXT: "OMG! WHR R U? LOL!".
Obvious so far. Here's where people might start to disagree with me...
What really surprises me about the iPhone is how little Apple got right. The following things also suck:
- The phone. There is a delay when picking up a new call and audio quality still sucks, regardless of method - ear buds, bluetooth, ear piece.
- No MMS for txt. Plain text SMS like it's 1999.
- Bluetooth. It's borderline useless. No profiles other than headset and A2DP (so I hear). Wow, I'm so glad my company paid so much for such a crippled device.
- The iPod. I don't HOW they screwed this one up... With just under 1,000 songs, the iPod software freezes the entire phone for a few seconds upon startup. Navigating around the various functions shows more of the same behavior - random freezes. The audio quality is horrible. HORRIBLE. Regardless of EQ settings or type of music, audio sounds weak and distorted. Don't ask me how that's possible at the same time, it just is. Any application of volume shows clipping and distortion, regardless of source material. The rotate function will drive you crazy. Listen to a song. Toss the iPhone in your pocket at any angle. When you pull it out the screen will be rotated horizontally in some kind of artist/album/song selection menu that scrolls around. Yes, it's very pretty but functionally, it's crap. Just like the rest of this device. God forbid you need to get out of this travesty and back to the somewhat functional vertically oriented interface. Good luck.
- Just like the iPod app, Contacts takes at least 3-5 seconds to start - EVERY TIME. I don't have that many contacts - just under 300 (297). I can't wait for a few seconds every time I need to interact with that database. I just can't.
- The calendar. This thing might work for you if all you have to do is set a reminder for the slumber party at your BFFs house this weekend. After all, you're a 13 old girl and that's all you have to worry about. That's obviously Apple's target market, and they NAILED it. Let's see - scroll wheels for setting dates and times are superfluous eye candy (more of the same from Apple). The alarm for events is useless. First of all, because it isn't persistent. It doesn't require you to acknowledge anything and rarely repeats. Secondly, it provides fixed options for notification before an event. I know I live 22 minutes away from a meeting. I want to set an alarm for 22 minutes. Too bad. I have an iPhone. Apple knows 13 year old girls don't need that and probably couldn't figure that interface out anyway. Setting schedules for recurring events - daily, weekly, etc. What if I want an event M-F, and another event M, W, F. How about another Saturday and Sunday. These recurring events are not supported. I have to create a new event for each day. Absolute crap. If I depended on this to make it to meetings and conference calls I'd be out of a job soon.
- The interface. I know I'm not the average user. I work (and play) in a command line environment all day and have for years. That means I'm used to having somewhat of an intimate connection with my devices. I type commands, hit enter, and it does what it is told. The man-machine communication is there. It's like driving a BMW. You really almost do feel one with the machine. I don't think I've ever felt so disconnected from a machine like I do with the iPhone. I never got the sense that I was really controlling this thing. From the keyboard to the excessive scrolling, rotating, blinking and other useless eye candy, to the lack of configurable options in everything I've come to the conclusion that I can't interface with this thing.
- Rotate. Make rotate system wide or don't support it at all. Very few apps actually support rotate. Again, I don't know what Apple was thinking here.
I've got plenty more but I'm done complaining for now. I came from a Nokia E-70. Let me tell you why this phone is awesome:
- Real keyboard. I can smoke ANYONE typing on this thing. Seriously, try me. Did I mention it also has a full phone keypad?
- Full bluetooth. Transfer files, push contacts, tether to your laptop. ANYTHING you want. It's even the little things... For instance - Symbian supports "profiles". Profiles allow you to configure just about any alert or setting on the device. You store them in profiles and can change carious settings with a single button. Think of memory seats in your car. It's basically that for phone settings. Anyway, Symbian supports automatically changing the profile based on which bluetooth device it's connected with. Basically, I used this to change the settings on my phone once it was connected to my car. I can never feel vibrate in my car for some reason. I can hear the tones I've assigned to this profile. I don't need to think about it. Get in the car, settings change on the phone. Get out, and they revert back to what they were before. Brilliant.
- PIM functions (Contacts, Calendar) have never done me wrong in over a year of ownership.
- MMS. Good MMS too.
- Awesome camera. 2MP, complete with night mode and video.
- 3G, 802.11g wifi, SIP.
- Real battery. Expandable storage.
Did I mention this phone is over two years old?
I'm serious about the iPhone being a play toy for teenagers. Why do you think it comes with Youtube? Once you're done playing Light Saber get a real phone - Blackberry or something running Symbian. That's it and that's why my iPhone SIM is in my E-70 and my iPhone is sitting on the table where I dump all of my junk mail when I walk in the door.
Seriously, if I didn't need this for my project I'd prop it up somewhere and unload some buckshot on it. Or maybe I'd give it to my friend Jason. He teaches middle school after all. Maybe he can sell it to one of his students and get this thing where it belongs.
This is the ONLY time you will find any profanity on this blog. Get an iPhone, and you'll see how justified it is:
The iPhone is a piece of shit, and so is your face.
Maddox, you are the man!
This time last week I was plotting how I could sneak into line to get one of those new iPhones. No it's not because I'm some Apply fanboy or some moron that can't do basic math (I'm not paying for the phone or the service). It's because my company is excited about what we might be able to do with the Apps store and the iPhone SDK. Trust me, it's really cool. That's all I can say right now.
I knew we needed at least a couple of these things (iPhones) and I didn't want to get stuck without them. The app we want to do is awesome and I wanted to get to work on it. That meant I was getting an iPhone. I had two choices:
1) Carry two phones
2) Port my number to our AT&T corporate plan and cancel T-Mobile
I wasn't going to carry two phones. A guy I work with carries at least two and it seems like a hassle. I HATE hassle. So it looks like I was going to port my number to AT&T and leave T-Mobile.
I'll skip the next part. We all know what happened here. Quick summary:
- In line at 7:45am
- Out of store by 10:30am (16GB white)
- Phone activated by 3:30pm
Let's just say I couldn't believe the activation problems. I guess Apple and AT&T were too busy marketing the hell out of this thing to remember to buy more servers or upgrade their network. Unbelievable.
Now that I've had the iPhone 3G for almost a week, here's what I've learned. Looking around the web, everyone already knows this:
- The camera sucks. Stills are grainy and often blurry. It can't do video at all. The Motorola RAZR could do video - five years ago. Today any free phone with a service plan can do video.
- Battery life sucks. For the first time in years, I actually worry about my cell phone battery.
- I don't care what anyone says, the keyboard sucks. It does ok considering there are no buttons but Apple needs to learn when they're wrong. They were wrong with the first gen, and they still are. It is not usable for any real work. I almost got to the point where I started sending incomprehensible teenage TXT: "OMG! WHR R U? LOL!".
Obvious so far. Here's where people might start to disagree with me...
What really surprises me about the iPhone is how little Apple got right. The following things also suck:
- The phone. There is a delay when picking up a new call and audio quality still sucks, regardless of method - ear buds, bluetooth, ear piece.
- No MMS for txt. Plain text SMS like it's 1999.
- Bluetooth. It's borderline useless. No profiles other than headset and A2DP (so I hear). Wow, I'm so glad my company paid so much for such a crippled device.
- The iPod. I don't HOW they screwed this one up... With just under 1,000 songs, the iPod software freezes the entire phone for a few seconds upon startup. Navigating around the various functions shows more of the same behavior - random freezes. The audio quality is horrible. HORRIBLE. Regardless of EQ settings or type of music, audio sounds weak and distorted. Don't ask me how that's possible at the same time, it just is. Any application of volume shows clipping and distortion, regardless of source material. The rotate function will drive you crazy. Listen to a song. Toss the iPhone in your pocket at any angle. When you pull it out the screen will be rotated horizontally in some kind of artist/album/song selection menu that scrolls around. Yes, it's very pretty but functionally, it's crap. Just like the rest of this device. God forbid you need to get out of this travesty and back to the somewhat functional vertically oriented interface. Good luck.
- Just like the iPod app, Contacts takes at least 3-5 seconds to start - EVERY TIME. I don't have that many contacts - just under 300 (297). I can't wait for a few seconds every time I need to interact with that database. I just can't.
- The calendar. This thing might work for you if all you have to do is set a reminder for the slumber party at your BFFs house this weekend. After all, you're a 13 old girl and that's all you have to worry about. That's obviously Apple's target market, and they NAILED it. Let's see - scroll wheels for setting dates and times are superfluous eye candy (more of the same from Apple). The alarm for events is useless. First of all, because it isn't persistent. It doesn't require you to acknowledge anything and rarely repeats. Secondly, it provides fixed options for notification before an event. I know I live 22 minutes away from a meeting. I want to set an alarm for 22 minutes. Too bad. I have an iPhone. Apple knows 13 year old girls don't need that and probably couldn't figure that interface out anyway. Setting schedules for recurring events - daily, weekly, etc. What if I want an event M-F, and another event M, W, F. How about another Saturday and Sunday. These recurring events are not supported. I have to create a new event for each day. Absolute crap. If I depended on this to make it to meetings and conference calls I'd be out of a job soon.
- The interface. I know I'm not the average user. I work (and play) in a command line environment all day and have for years. That means I'm used to having somewhat of an intimate connection with my devices. I type commands, hit enter, and it does what it is told. The man-machine communication is there. It's like driving a BMW. You really almost do feel one with the machine. I don't think I've ever felt so disconnected from a machine like I do with the iPhone. I never got the sense that I was really controlling this thing. From the keyboard to the excessive scrolling, rotating, blinking and other useless eye candy, to the lack of configurable options in everything I've come to the conclusion that I can't interface with this thing.
- Rotate. Make rotate system wide or don't support it at all. Very few apps actually support rotate. Again, I don't know what Apple was thinking here.
I've got plenty more but I'm done complaining for now. I came from a Nokia E-70. Let me tell you why this phone is awesome:
- Real keyboard. I can smoke ANYONE typing on this thing. Seriously, try me. Did I mention it also has a full phone keypad?
- Full bluetooth. Transfer files, push contacts, tether to your laptop. ANYTHING you want. It's even the little things... For instance - Symbian supports "profiles". Profiles allow you to configure just about any alert or setting on the device. You store them in profiles and can change carious settings with a single button. Think of memory seats in your car. It's basically that for phone settings. Anyway, Symbian supports automatically changing the profile based on which bluetooth device it's connected with. Basically, I used this to change the settings on my phone once it was connected to my car. I can never feel vibrate in my car for some reason. I can hear the tones I've assigned to this profile. I don't need to think about it. Get in the car, settings change on the phone. Get out, and they revert back to what they were before. Brilliant.
- PIM functions (Contacts, Calendar) have never done me wrong in over a year of ownership.
- MMS. Good MMS too.
- Awesome camera. 2MP, complete with night mode and video.
- 3G, 802.11g wifi, SIP.
- Real battery. Expandable storage.
Did I mention this phone is over two years old?
I'm serious about the iPhone being a play toy for teenagers. Why do you think it comes with Youtube? Once you're done playing Light Saber get a real phone - Blackberry or something running Symbian. That's it and that's why my iPhone SIM is in my E-70 and my iPhone is sitting on the table where I dump all of my junk mail when I walk in the door.
Seriously, if I didn't need this for my project I'd prop it up somewhere and unload some buckshot on it. Or maybe I'd give it to my friend Jason. He teaches middle school after all. Maybe he can sell it to one of his students and get this thing where it belongs.
This is the ONLY time you will find any profanity on this blog. Get an iPhone, and you'll see how justified it is:
The iPhone is a piece of shit, and so is your face.
Maddox, you are the man!
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