Hi Markus, Can you elaborate upon the way in which you are using Kamailio+RTPEngine "as a drop-in replacement"? Drop-in replacement for what? Or that is to say, what are you trying to accomplish here, functionally?
I have the suspicion that what you're doing is probably best accomplished in a different and more straightforward way. :-) -- Alex > On Sep 21, 2023, at 8:58 PM, Markus <[email protected]> wrote: > > Hi list, > > I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted emergency > situation (didn't monitor traffic growth properly and now encountering packet > loss during peak times) as a drop-in replacement for an overloaded Asterisk > box in a call-termination-to-upstream-carrier scenario. > > My test scenario is to make a call from a SIP softphone to Asterisk IP > 1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3 > > sngrep on Kamailio box 2.2.2.2 - the following SDP will not work - carrier is > rejecting it. Carrier is authenticating our calls based on our IP address > 2.2.2.2, no username/pass involved. > > 2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060 > INVITE sip:[email protected];user=phone SIP/2.0 > Record-Route: <sip:2.2.2.2;lr> > Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0 > Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8 > Max-Forwards: 69 > From: "61xxxxxxxxx" <sip:[email protected]>;tag=as3d75aadd > To: <sip:[email protected];user=phone> > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > User-Agent: Asterisk PBX 20.0.0 > Date: Fri, 22 Sep 2023 00:06:50 GMT > Session-Expires: 1800 > Min-SE: 90 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > P-Asserted-Identity: <sip:[email protected];user=phone> > Content-Type: application/sdp > Content-Length: 314 > X-SIP: 1.1.1.1 > > v=0 > o=root 1093000903 1093000903 IN IP4 1.1.1.1 > s=Asterisk PBX 20.0.0 > c=IN IP4 2.2.2.2 > t=0 0 > m=audio 25742 RTP/AVP 8 9 0 101 > a=maxptime:150 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendrecv > a=rtcp:25743 > a=ptime:20 > > I'm comparing this rejected INVITE to a successful INVITE sent by the > original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier without > Kamailio in the path, and these are the differences I noticed, and probably > the things I have to mimick with Kamailio in order to make it work: > > INVITE sip:[email protected];user=phone SIP/2.0 > should be > INVITE sip:[email protected];user=phone SIP/2.0 > > To: <sip:[email protected];user=phone> > should be > To: <sip:[email protected];user=phone> > > From: "61xxxxxxxxx" <sip:[email protected]>;tag=as3d75aadd > should be > From: "61xxxxxxxxx" <sip:[email protected]>;tag=as3d75aadd > > Contact: <sip:[email protected]:5060> > should be > Contact: <sip:[email protected]:5060> > > Call-ID: [email protected]:5060 > should be > Call-ID: [email protected]:5060 > > o=root 1093000903 1093000903 IN IP4 1.1.1.1 > should be > o=root 1093000903 1093000903 IN IP4 2.2.2.2 > > My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU > > These are the Asterisk boxes I want to originate calls from to Kamailio: > > [root@voip30 ~]# kamctl address show > +-----+-----+----------+------+------+-----------+ > | id | grp | ip_addr | mask | port | tag | > +-----+-----+----------+------+------+-----------+ > | 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv | > | 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv | > | 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv | > | 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv | > | 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv | > | 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv | > | 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv | > | 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv | > | 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv | > +-----+-----+----------+------+------+-----------+ > > This is the upstream carrier I want Kamailio to proxy calls to: > > [root@voip30 ~]# kamctl dispatcher show > dispatcher gateways > +----+-------+------------------+-------+-------+------------+------+ > | id | setid | destination | flags | prio. | attrs | desc | > +----+-------+------------------+-------+-------+------------+------+ > | 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | | > +----+-------+------------------+-------+-------+------------+------+ > (output manually slightly modified to look properly over E-Mail) > > As you might have guessed I'm a Kamailio noob... and don't have the resources > to learn it as fast as I must to avoid further packet loss. If there's anyone > available who can help me to get this done today, optionally in exchange for > money, I'd be grateful. > > Thank you! > Markus > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to [email protected] > Important: keep the mailing list in the recipients, do not reply only to the > sender! > Edit mailing list options or unsubscribe: -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
