[asterisk-users] PJSIP signaling question

Fri Mar 4 14:45:55 CST 2016

I can’t quite figure it out , I went ahead and pulled everything yet again, and I made sure to delete everything related to pjproject from my system, all the PJ  lib and include files that were in /usr/lib/  ,  I pulled pjproject from svn , pulled asterisk code from gerrit, recompiled everything, but still I think new TLS transports are being made which fail in my NAT scenarios .  I check with: 

tcpdump -i any src host 10.50.55.10  and  'tcp[13] & 2 != 0’ 

I see tcpdump print a new tcp SYN packet when I  try to make a call between endpoints and also when Asterisk tries to send OPTIONS command to the endpoint .

From my endpoints, I can call the “echo” applications and the call works fine, but I cannot call from one endpoint to another endpoint , even though they are both egistered. It does not say “unavailable’ or anything,  I see in the pjsip log that an INVITE is  “sent” , but I think the logger is just showing me that the INVITE message has been created, but it never reaches the endpoint because of the new TLS connection failing because of the NAT. Eventually, the call times out with a 408 error in the pjsip log.

I also see some log entries:
[Mar  4 12:29:10] DEBUG[16225] pjsip: 	tlsc0x7f311400 TLS connect() error: Connection timed out [code=120110]
[Mar  4 12:29:29] DEBUG[16225] pjsip: 	tlsc0x7f311400 TLS connect() error: Connection timed out [code=120110]

Just to be clear I am getting pjproject like so : 
svn co http://svn.pjsip.org/repos/pjproject/trunk

and asterisk :
git clone -b 13 http://gerrit.asterisk.org/asterisk

then I go to pjproject directory,  create a site_config.h file (to increase TLS connectors and set other options recommended on Wiki)

configure pjproject with the following options:

./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp

Then go to asterisk directory

make clean; make distclean; ./boostrap.sh ; ./configure;  make menuselect; make; make install;

> On Mar 4, 2016, at 7:33 AM, George Joseph <george.joseph at fairview5.com> wrote:
> 
> 
> 
> On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <kevin.long at haloprivacy.com> wrote:
> Hi George the patch was from here , you wrote it I believe . I pulled asterisk 13 from git, apply this patch which fixed RTP issue , but I think tla transport issue came back for me . 
> 
> https://gerrit.asterisk.org/#/c/2346/
> 
> Oh, that one, OK.    It should be merged now so if you 'git pull' on 13 now, you should get it.  The transport re-use issue was in pjproject so is it possible that you're not compiling against the latest trunk?
> 
> 
> 
> 
>  
> 
> Thank you
> 
> Sent from my iPhone
> 
> On Mar 4, 2016, at 12:01 AM, George Joseph <george.joseph at fairview5.com> wrote:
> 
>> 
>> 
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long at haloprivacy.com> wrote:
>> 
>> Thanks George I appreciate the info .  Being able to see what codec is in use for call in progress is very handy sometimes.
>> 
>> As far as the RTP stats goes,  I see there is some info with “rtp” and “rtcp” commands which can be useful for troubleshooting. A running tally of # packets or bandwidth used would be awesome in along with the codec in "pjsip show channels" or something like that.
>> 
>> 
>> Im not certain, but I think the TLS signalling problem from this email may be happening to me again after patching for another pjsip/NAT issue which was with the external_media_address not working and the internal IP being sent in the SDP from asterisk - I applied this patch to the codebase and recompiled I am seeing the TLS “new transport”  issue again , I think.
>> 
>> I've lost track of who's applying what patches to which codebase. :)
>> 
>> Which patch did you apply for "external_media_address not working"?
>> 
>>  
>> 
>> Regards,
>> 
>> Kevin Long
>> --
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