[jitsi-dev] Question/Observation about SC


#1

I use SIP Phones in general a lot to make direct SIP calls. I don't need to have a host to make one of these calls, just specify something like sip:200901@login.zipdx.com to connect to the VOIP Users Conference on Fridays at noon (You can call in too if you like)... It appears to me that SIP Communicator wants me to set up an account that SC will register to and then contact through the host. It's not intuitively obvious to me the best or easiest way to set up my weekly ZipDX VUC call in the phone book.

b) When I make the ZipDX call each friday with SC, it works for about 2 to 3 minutes and then the call drops. I contacted ZipDX about this (David Frankel-dfrankel@zipdx.com / 1-800-372-6535). He says _"I looked at our logs. Your client is not responding to the SIP UPDATE messages that we send periodically. Even if a client doesn't support a particular method, it still needs to respond (with, for example, "405 METHOD NOT ALLOWED"). The client's failure to respond (after many retries) makes us assume that it has gone away, so we disconnect."_

Item B makes SIP Communicator almost unusable (have to reconnect every 2 to 3 minutes) in this setting. I'm assuming that the ZipDX people deal with a lot of SIP dialers and know what they are talking about, but I've put the ZipDX contact persons info in the message in case you want to compare notes.

Happy Hunting.

···

--
John F. Ervin
Central Florida TeleSource
407-679-6238
http://jervin.com/cft
jervin@jervin.com


#2

Hey John,

На 03.02.11 21:03, John Ervin написа:

I use SIP Phones in general a lot to make direct SIP calls. I don't
need to have a host to make one of these calls, just specify something
like sip:200901@login.zipdx.com to connect to the VOIP Users Conference
on Fridays at noon (You can call in too if you like)... It appears to
me that SIP Communicator wants me to set up an account that SC will
register to and then contact through the host.

Although, for a variety of reasons, this would definitely be the best
way to connect, SC does not require you to use a server. In order to
make calls without using a proxy you need to create what we call a
Registrarless account. To do so, create a new SIP account and only enter
a user name without a server part. You should see an example in the
account wizard. In other words, if you are determined to run without a
server then instead of setting your account as "jervin@example.com" you
should configure it as simply "jervin".

b) When I make the ZipDX call each friday with SC, it works for about
2 to 3 minutes and then the call drops.

Strange. I often connect to VUC (using SC of course) and never had the
issue. My last call was more than a month ago however so something may
have changed since then.

One thing we changed recently and that may be related is the fact that,
for security reasons, we started dropping requests that arrive from a
server that we are not connected to. Of course we only do this for
accounts that actually have servers.

I contacted ZipDX about this
(David Frankel-dfrankel@zipdx.com / 1-800-372-6535). He says _"I looked
at our logs. Your client is not responding to the SIP UPDATE messages
that we send periodically. Even if a client doesn't support a
particular method, it still needs to respond (with, for example, "405
METHOD NOT ALLOWED"). The client's failure to respond (after many
retries) makes us assume that it has gone away, so we disconnect."_

I just had a call with wbdemo@conf.zipdx.com and got dropped exactly 10
minutes after I started the call. I am not sure whether this was the
same problem as the one you encountered since it could just be about
how wbdemo is configured.

However, I didn't receive a single SIP packet from zipdx between the
initial INVITE transaction and the BYE request. In other words there was
nothing for us to ignore.

I won't be able to join tomorrow's VUC because of FOSDEM so if you or
anyone else could be there and sees the problem, then it would be nice
to see the logs (and if posible pcap dumps).

Cheers,
Emil

···

Item B makes SIP Communicator almost unusable (have to reconnect every 2
to 3 minutes) in this setting. I'm assuming that the ZipDX people deal
with a lot of SIP dialers and know what they are talking about, but I've
put the ZipDX contact persons info in the message in case you want to
compare notes.

Happy Hunting.

--
John F. Ervin
Central Florida TeleSource
407-679-6238
http://jervin.com/cft
jervin@jervin.com

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31


#3

o John Ervin on 02/03/2011 09:03 PM:

  I use SIP Phones in general a lot to make direct SIP calls. I don't need to have a host to make one of these calls, just specify something like sip:200901@login.zipdx.com to connect to the VOIP Users Conference on Fridays at noon (You can call in too if you like)... It appears to me that SIP Communicator wants me to set up an account that SC will register to and then contact through the host. It's not intuitively obvious to me the best or easiest way to set up my weekly ZipDX VUC call in the phone book.

b) When I make the ZipDX call each friday with SC, it works for about 2 to 3 minutes and then the call drops. I contacted ZipDX about this (David Frankel-dfrankel@zipdx.com / 1-800-372-6535). He says _"I looked at our logs. Your client is not responding to the SIP UPDATE messages that we send periodically. Even if a client doesn't support a particular method, it still needs to respond (with, for example, "405 METHOD NOT ALLOWED"). The client's failure to respond (after many retries) makes us assume that it has gone away, so we disconnect."_

to me this sounds much more like you are behind a NAT and zipdx can't send any message to you (either NAT binding on SIP port expired, or pivate IP in contact). Check your log whether in sc you see any message from zipdx. that's also a good reason to use a proxy...

Stefan


#4

what are you using to log. I will be happy to log my session for a while.

BTW, I had the same thing happen when I was testing. It dropped at 10 minutes. David Frankel says: "Yes, there is a difference. The demo (3366) does NOT use SIP UPDATEs to periodically "validate" your call. The demo has an explicit time-out at 10 minutes. I believe that if you dial 923366, it will SEND the updates -- so you could at least capture what was being sent back and forth. However, even with 923366, I don't think it will hang up on you if it doesn't get a response."

So, can you retry your test using 923366@login.zipdx.com? It still doesn't kick you off, but it does do the SIP Updates so we can test it"

···

I just had a call with wbdemo@conf.zipdx.com and got dropped exactly 10
minutes after I started the call. I am not sure whether this was the
same problem as the one you encountered since it could just be about
how wbdemo is configured.

However, I didn't receive a single SIP packet from zipdx between the
initial INVITE transaction and the BYE request. In other words there was
nothing for us to ignore.

I won't be able to join tomorrow's VUC because of FOSDEM so if you or
anyone else could be there and sees the problem, then it would be nice
to see the logs (and if posible pcap dumps).

Cheers,
Emil


#5

I don't think I've ever used a proxy for a SIP call. I don't have the problem with any my other SIP Phones. I just have other problems, like no G722 or doesn't work well with Ubuntu/PulseAudio. SIP Communicator so far seems like a good combination of features and working well with Ubuntu 10.10

···

On 02/03/2011 05:54 PM, Stefan Sayer wrote:

o John Ervin on 02/03/2011 09:03 PM:

  I use SIP Phones in general a lot to make direct SIP calls. I don't need to have a host to make one of these calls, just specify something like sip:200901@login.zipdx.com to connect to the VOIP Users Conference on Fridays at noon (You can call in too if you like)... It appears to me that SIP Communicator wants me to set up an account that SC will register to and then contact through the host. It's not intuitively obvious to me the best or easiest way to set up my weekly ZipDX VUC call in the phone book.

b) When I make the ZipDX call each friday with SC, it works for about 2 to 3 minutes and then the call drops. I contacted ZipDX about this (David Frankel-dfrankel@zipdx.com / 1-800-372-6535). He says _"I looked at our logs. Your client is not responding to the SIP UPDATE messages that we send periodically. Even if a client doesn't support a particular method, it still needs to respond (with, for example, "405 METHOD NOT ALLOWED"). The client's failure to respond (after many retries) makes us assume that it has gone away, so we disconnect."_

to me this sounds much more like you are behind a NAT and zipdx can't send any message to you (either NAT binding on SIP port expired, or pivate IP in contact). Check your log whether in sc you see any message from zipdx. that's also a good reason to use a proxy...

Stefan

--
John F. Ervin
Central Florida TeleSource
407-679-6238
http://jervin.com/cft
jervin@jervin.com


#6

Hey John,

На 03.02.11 22:40, John Ervin написа:

what are you using to log. I will be happy to log my session for a while.

BTW, I had the same thing happen when I was testing. It dropped at 10
minutes. David Frankel says: "Yes, there is a difference. The demo
(3366) does NOT use SIP UPDATEs to periodically "validate" your call.
The demo has an explicit time-out at 10 minutes. I believe that if you
dial 923366, it will SEND the updates -- so you could at least capture
what was being sent back and forth. However, even with 923366, I don't
think it will hang up on you if it doesn't get a response."

So, can you retry your test using 923366@login.zipdx.com? It still
doesn't kick you off, but it does do the SIP Updates so we can test it"

OK, reproduced and repaired. Build 3272 (currently baking) should have
the fix.

Thanks for your report!

Cheers,
Emil

···

I just had a call with wbdemo@conf.zipdx.com and got dropped exactly 10
minutes after I started the call. I am not sure whether this was the
same problem as the one you encountered since it could just be about
how wbdemo is configured.

However, I didn't receive a single SIP packet from zipdx between the
initial INVITE transaction and the BYE request. In other words there was
nothing for us to ignore.

I won't be able to join tomorrow's VUC because of FOSDEM so if you or
anyone else could be there and sees the problem, then it would be nice
to see the logs (and if posible pcap dumps).

Cheers,
Emil

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31


#7

На 03.02.11 23:54, Stefan Sayer написа:

o John Ervin on 02/03/2011 09:03 PM:

  I use SIP Phones in general a lot to make direct SIP calls. I don't
need to have a host to make one of these calls, just specify something
like sip:200901@login.zipdx.com to connect to the VOIP Users Conference
on Fridays at noon (You can call in too if you like)... It appears to
me that SIP Communicator wants me to set up an account that SC will
register to and then contact through the host. It's not intuitively
obvious to me the best or easiest way to set up my weekly ZipDX VUC call
in the phone book.

b) When I make the ZipDX call each friday with SC, it works for about
2 to 3 minutes and then the call drops. I contacted ZipDX about this
(David Frankel-dfrankel@zipdx.com / 1-800-372-6535). He says _"I looked
at our logs. Your client is not responding to the SIP UPDATE messages
that we send periodically. Even if a client doesn't support a
particular method, it still needs to respond (with, for example, "405
METHOD NOT ALLOWED"). The client's failure to respond (after many
retries) makes us assume that it has gone away, so we disconnect."_

to me this sounds much more like you are behind a NAT and zipdx can't
send any message to you (either NAT binding on SIP port expired, or
pivate IP in contact). Check your log whether in sc you see any
message from zipdx. that's also a good reason to use a proxy...

Oh yes, I did want to mention this: other than a couple of UPDATEs in
the beginning of the call, zipdx doesn't seem to send any other
keepalives so if you are not using a proxy and are behind a NAT, you
will simply stop getting SIP messages from them at some point (even if
RTP would probably continue).

Setting up an account on a public SIP server (such as iptel.org or
ippi.com for example) is a 30 second matter so there's really no reason
to avoid it. Quite on the contrary.

Cheers,
Emil


#8

Confirmed that SIP Communicator is no longer getting dropped by ZipDX at 2 to 3 minutes. I have been on for about 20 minutes now. Good work...

···

--
John F. Ervin
Central Florida TeleSource
407-679-6238
http://jervin.com/cft
jervin@jervin.com