Hello, I have an asterisk setup using pjsip. Everything used to work correctly until one remote site changed internet provider and thier router does not support sip protocol algorithms. It works for some time, but then suddenly audio stops working both directions. When this happens I see RTP responses going out to the local address of the natted phone, not to the natted address. The problem appears for the phones independently. The asterisk is connected to the internet with public static IP address. The pjsip config contains: [aor] type=aor qualify_frequency = 60 max_contacts=1 remove_existing = yes [endpoint] type = endpoint context = internal dtmf_mode = rfc4733 disallow = all allow = alaw allow = ilbc allow = g729 allow = gsm allow = g723 direct_media = no allow_subscribe = yes subscribe_context = blf rewrite_contact = yes rtp_symmetric = yes force_rport = yes Am I missing something? Why the communication breaks suddenly? Thanks Marek
Have you tried to see if the router has any SIP ALG enabled this might create such issues, Thanks. Best Regards, On Thu, Jul 8, 2021, 19:59 Marek Greško <mgresko8 at gmail.com> wrote:> Hello, > > I have an asterisk setup using pjsip. Everything used to work > correctly until one remote site changed internet provider and thier > router does not support sip protocol algorithms. > > It works for some time, but then suddenly audio stops working both > directions. When this happens I see RTP responses going out to the > local address of the natted phone, not to the natted address. The > problem appears for the phones independently. > > The asterisk is connected to the internet with public static IP address. > > The pjsip config contains: > > [aor] > type=aor > qualify_frequency = 60 > max_contacts=1 > remove_existing = yes > > [endpoint] > type = endpoint > context = internal > dtmf_mode = rfc4733 > disallow = all > allow = alaw > allow = ilbc > allow = g729 > allow = gsm > allow = g723 > direct_media = no > allow_subscribe = yes > subscribe_context = blf > rewrite_contact = yes > rtp_symmetric = yes > force_rport = yes > > > Am I missing something? Why the communication breaks suddenly? > > Thanks > > Marek > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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/20210708/4f42f810/attachment.html>
El jue, 8 de jul. de 2021 a la(s) 14:58, Marek Greško (mgresko8 at gmail.com) escribió:> The asterisk is connected to the internet with public static IP address. > > The pjsip config contains: > >What does your transport config look like? Take a look at this wiki page: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT -- Michael L. Young (elguero) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20210708/7126e7c3/attachment.html>
On Thursday 08 July 2021 at 20:57:30, Marek Greško wrote:> Hello, > > I have an asterisk setup using pjsip. Everything used to work > correctly until one remote site changed internet provider and thier > router does not support sip protocol algorithms.I'm slightly bothered by the word "algorithms" in that comment, but I do wonder whether it simply means that this is a connectivity provider (possibly a mobile phone network?) which actively blocks SIP. Some of them (in my experience) do this by blocking UDP port 5060, but others are more subtle about it, and (for example) block the authentication responses to a Register request, thereby meaning that UDP port 5060 is in general accessible, but any attempt to Register to it using SIP will fail. Have you asked the new Internet connectivity provider whether they support or block SIP across their network? Antony -- "Remember: the S in IoT stands for Security." - Jan-Piet Mens Please reply to the list; please *don't* CC me.