asterisk users - Feb 2016 - Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Wow. Incredible. That worked. The backslash is important there; I kept
> trying with no backslash and followed the instructions in
> pjsip_wizard.conf.sample (in configs/samples) and it says we have to say
>
> transport=tcp ; the only example however talks about ipv4.
>
> Is this documented somewhere and I just missed it??
>
> So, let me sum the issues and their solutions:
>
> (a) Inside/from-internal calling. Only need transport=tcp in pjsip.conf.
> No need to update every SIP (user) endpoint's transport, though that
did
> not disrupt anything.
> (b) For pjsip_wizard configuration, add the transport into the
> remote_hosts line like so noting that the backslash is important otherwise
> the transport part of the line is a comment!
>
> remote_hosts = silly.pstn.twilio.com?\;transport=tcp
>
> Simple errors, but vexing, vexing, vexing issues.
>
One thing to be aware of...?

There is currently a PJSIP bug when using TCP and TLS that shows up if you
explicitly
set transport= on an endpoint (or in the wizard).  It's best to leave
transport unset and
let PJSIP determine the transport from the ;transport= parameter of the URI.
>From a wizard perspective, if you have lots of TCP or TLS endpoints, use a
template like so...

[tcp-template](!)
server_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP
client_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP
contact_pattern = sip:${REMOTE_HOST}\;transport=TCP

[tls-template](!)
server_uri_pattern = sips:${REMOTE_HOST}\;transport=TLS
client_uri_pattern = sips:${REMOTE_HOST}\;transport=TLS
contact_pattern = sips:${REMOTE_HOST}\;transport=TLS

[tcp-provider](tcp-template]
remote_hosts = my.provider.net

Let me know if the wiki can use some clarification.  I haven't updated it
in a while.

>
> Thanks, George, and thanks Joshua, for your time!
>
>
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George,

May I propose we improve the documentation on the Asterisk Wiki? I thought
I would have spent far less time here (though you folks have been mightily
helpful, and thanks again!) should the documentation for the TCP transport
be improved in both the Wiki and specifically, in
${ASTERISK_HOME}/configs/samples/, bundled as part of the code. I want to
see Asterisk as a product succeed (even more) and Asterisk in its new
version succeed wildly.

I don't know if you folks allow outside developers to pitch in, but
depending on a number of factors, I might contribute to
https://gerrit.asterisk.org/ if that is within Asterisk's policy. Again,
depending on a number of factors, including legal.

Here, specifically, is the list of improvements I propose:

(a) One full example showing how a TCP based Asterisk platform should work
in the PJSIP world, including both SIP over TCP-compliant SIP trunk
configuration using pjsip_wizard.conf
(b) One complete example reflected in the distributed code samples within
Asterisk code, in ${ASTERISK_HOME}/configs/samples/.
(c) A full SIP trace for Asterisk, the working examples, for all manner of
transports (UDP, TCP, TLS), for REGISTER, INVITE etc.

Hope this helps.

Thanks again!

On Wed, Feb 17, 2016 at 3:48 PM, George Joseph <george.joseph at
fairview5.com>
wrote:
>
>
> On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> Wow. Incredible. That worked. The backslash is important there; I kept
>> trying with no backslash and followed the instructions in
>> pjsip_wizard.conf.sample (in configs/samples) and it says we have to
say
>>
>> transport=tcp ; the only example however talks about ipv4.
>>
>> Is this documented somewhere and I just missed it??
>>
>> So, let me sum the issues and their solutions:
>>
>> (a) Inside/from-internal calling. Only need transport=tcp in
pjsip.conf.
>> No need to update every SIP (user) endpoint's transport, though
that did
>> not disrupt anything.
>> (b) For pjsip_wizard configuration, add the transport into the
>> remote_hosts line like so noting that the backslash is important
otherwise
>> the transport part of the line is a comment!
>>
>> remote_hosts = silly.pstn.twilio.com?\;transport=tcp
>>
>> Simple errors, but vexing, vexing, vexing issues.
>>
>
> One thing to be aware of...?
>
> There is currently a PJSIP bug when using TCP and TLS that shows up if you
> explicitly
> set transport= on an endpoint (or in the wizard).  It's best to leave
> transport unset and
> let PJSIP determine the transport from the ;transport= parameter of the
> URI.
>
> From a wizard perspective, if you have lots of TCP or TLS endpoints, use a
> template like so...
>
> [tcp-template](!)
> server_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP
> client_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP
> contact_pattern = sip:${REMOTE_HOST}\;transport=TCP
>
> [tls-template](!)
> server_uri_pattern = sips:${REMOTE_HOST}\;transport=TLS
> client_uri_pattern = sips:${REMOTE_HOST}\;transport=TLS
> contact_pattern = sips:${REMOTE_HOST}\;transport=TLS
>
> [tcp-provider](tcp-template]
> remote_hosts = my.provider.net
>
> Let me know if the wiki can use some clarification.  I haven't updated
it
> in a while.
>
>
>>
>> Thanks, George, and thanks Joshua, for your time!
>>
>>
>
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Sonny Rajagopalan wrote:
> George,
>
> May I propose we improve the documentation on the Asterisk Wiki? I
> thought I would have spent far less time here (though you folks have
> been mightily helpful, and thanks again!) should the documentation for
> the TCP transport be improved in both the Wiki and specifically, in
> ${ASTERISK_HOME}/configs/samples/, bundled as part of the code. I want
> to see Asterisk as a product succeed (even more) and Asterisk in its new
> version succeed wildly.
>
> I don't know if you folks allow outside developers to pitch in, but
> depending on a number of factors, I might contribute to
> https://gerrit.asterisk.org/ if that is within Asterisk's policy.
Again,
> depending on a number of factors, including legal.
Anyone can contribute changes to Asterisk provided a contributor license 
agreement is signed. The process is on the wiki itself too[1].

[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org