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 > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160304/49f61496/attachment-0001.html>
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/ 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 >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160304/991cb1f4/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2384 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160304/991cb1f4/attachment.bin>
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 >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160304/76119106/attachment.html>