tag:blogger.com,1999:blog-7979378259821020425.post6002914602935650224..comments2024-01-31T05:01:09.874-08:00Comments on SaevolGo: RTPproxy Revisited [Kamailio 4.0]Unknownnoreply@blogger.comBlogger29125tag:blogger.com,1999:blog-7979378259821020425.post-20403406686354559202017-01-03T22:30:25.946-08:002017-01-03T22:30:25.946-08:00Sounds like an ambitious plan for a huge audio con...Sounds like an ambitious plan for a huge audio conf system. <br />Well again, share how that Page(multicastrtp) is dialled form asterisk ?<br /><br />kamailio is only going to get the INVITE find where it needs to be routed to and relay it there. the SDP if needs to be modified for RTPproxy or not is again a configurable area for this.Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-11448904017592297802017-01-03T21:53:28.898-08:002017-01-03T21:53:28.898-08:00user1---joins--->(asterisk1 conference)---Page(...user1---joins--->(asterisk1 conference)---Page(multicastrtp)--->multicast group------->(asterisk2 conference room 2000 + asterisk3 conference room 3000 +asterisk4 conference 4000)<br /><br />the above is my scenario, in simple word i need to send multicast rtp from one conference (asterisk) to n-1 asterisk conference rooms.<br /><br />Please let me know how we can approach this with the help of kamailio and asterisk<br /><br /><br />the above Anonymoushttps://www.blogger.com/profile/08546725562621654568noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-33366417985402116982017-01-03T21:29:55.412-08:002017-01-03T21:29:55.412-08:00Thats definitely possible but extar code would be ...Thats definitely possible but extar code would be required such that kamailio at the initial invite understands that rtpproxy dont need to be engaged since RTPs are flowing within the "Asterisk" subnet directly. Similarly the INVITE needs to be sent to the target asterisk(s)><br /><br />in simple words "it depends on the call flow scenario" Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-88548635917604379362017-01-03T21:22:06.499-08:002017-01-03T21:22:06.499-08:00Hi this is such a great Blog,
i am new to Kamailio...Hi this is such a great Blog,<br />i am new to Kamailio, but do have experience in asterisk. if i am doing a multicast paging to Kamailio from one asterisk server, is it possible to send this rtp to multiple asterisk servers via SIP channels?Anonymoushttps://www.blogger.com/profile/08546725562621654568noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-32682734565496819482016-07-18T14:07:56.442-07:002016-07-18T14:07:56.442-07:00I couldn't understand the problem well, kindly...I couldn't understand the problem well, kindly share some personal contact info so I can hear you out and make some recommendations.Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-14909126793616238572016-07-18T12:25:25.168-07:002016-07-18T12:25:25.168-07:00Configuración de proxy (servidor VoIP de contrato)...Configuración de proxy (servidor VoIP de contrato) en ruters con fxo.<br /><br />Hola, como puedo especificar la dirección de la llamadas, si me llaman por el proxy las puedo mandar a los fxs, he visto que a veces el srp cuando recibe llamadas hace peticiones sip y rtp, lo que no se es por que no puedo identificar que pasa, creo que si contrato un número IP puedo hacer sonar los fxs pero desconozco que pasará con esos paquetes sip, creo que hay configuracion sip en el pstn y enable ip equivale a los sipura voip to pstn o algo así pero cada vez que consigo hacer algo el equipo pierde configuraciones o tal vez esté pasando pr varias centrales en el fxo y por esto no acaba de funcionar el adaptador del IOS cisco y no se volver a la primera central, o algo así me he esperado pero nada no detecta el fallo y no va, sobre todo suele perder tonos en el fxs y llamadas entrantes, y a veces el tono en la fxo.<br />El proxy asterisk (servidor en mac) me enseña esto, bueno te deja comprobar funcionalidades para mandar llamadas entre oficinas o ver que le pasa a tu telefono SIP.<br /><br />SIP/2.0 200 OK<br />To: ;tag=afa6e570d2160c87i0<br />From: "asterisk" ;tag=as482fd221<br />CALL-ID: 44ccef1a116545094de6154d49b71fdd@192.168.1.105:5060<br />CSeq: 102 OPTIONS<br />Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK064d78f0<br />Server: Cisco/SRP547-1.2.6(003)<br />Content-Length: 0<br />Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<br />Supported: x-sipura, replaces<br />Accept-Language: en<br /><br /><br />GREAT DAY!!Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-7572377556558548922015-06-26T03:53:54.827-07:002015-06-26T03:53:54.827-07:00Hello,
We are having the same problem as you descr...Hello,<br />We are having the same problem as you described here. Did you find a solution?<br />Thanks,<br />EmmanuelAnonymoushttps://www.blogger.com/profile/15307869290634160046noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-59867859004220898612015-06-25T19:02:24.702-07:002015-06-25T19:02:24.702-07:00Hello. Were ever able to solve this?Hello. Were ever able to solve this?Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-8538566132491874892015-06-25T19:01:09.946-07:002015-06-25T19:01:09.946-07:00Hi dipak,
We are having same issue. Were you able ...Hi dipak,<br />We are having same issue. Were you able to resolve it? Would you care to share how?Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-37276057479743945702015-01-15T00:45:58.620-08:002015-01-15T00:45:58.620-08:00Kindly can you pls guide with the below
Kamailio ...Kindly can you pls guide with the below<br /><br />Kamailio - 4.2.2 ( SIP server )<br />Rtpproxy - Git Compiled ( miconda patched version )<br /><br />Issue:<br />Remote NAT Call<br />Bria Rmt Iphone SIP Extn (3G) ----> Kamailio Server -----> Desktop Bria Client ( Wifi )<br /><br />Audio and Video packets are sent from iPhone to desktop client .. but nothing otherway<br /><br />Thanks<br />Chirag A<br />chirag_ja@yahoo.co.inAnonymousnoreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-20955161312760121282014-10-21T08:34:37.861-07:002014-10-21T08:34:37.861-07:00Everything look fine in terms of RTPproxy configs,...Everything look fine in terms of RTPproxy configs, there is no media coming from any of the side and what Im finding interesting is that the originator of this call has set SDP attricbite "a=receive only".<br />Try changing your softphone, also see if there are any firewalls on your Kamailio server or at the either legs.<br /><br />which IP you're not sure about. Need to know more. Contact me on private email plz.Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-68076221938301652882014-10-21T08:13:00.201-07:002014-10-21T08:13:00.201-07:00Hi
Thanks for all the details, I currently have Ka...Hi<br />Thanks for all the details, I currently have Kamailio 4.1 installed, I have tried to paste the RTPProxy control section in to my cfg file, however service could not start.<br />Currently I can make calls however I have no voice on both sides.<br />I got a PCAP that I can share, there is one strange IP in pcap that I am not sure about.<br />Any hints for why I can not hear anything or paste the section above to my cfg ?<br />link to pcap https://www.dropbox.com/s/d6l1h34w6z54kkm/rtp-calls.zip?dl=0 <br />Anonymoushttps://www.blogger.com/profile/12540403286373877203noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-41402051746327333252014-07-01T02:59:53.132-07:002014-07-01T02:59:53.132-07:00Hello, I trying to integrate this scheme on my dep...Hello, I trying to integrate this scheme on my deploy. My problem is sending valid ACK from called party to Asterisk through Kamailio. When I recieve OK message Contact header contains private IP of asterisk server, so ACK try sends back to private ip of asterisk server. Offcourse it is wrong, because private ip is reacheble only through Kamailio.<br /><br />I've rtied this:<br />onreply_route[MANAGE_REPLY] {<br /> xdbg("incoming reply\n");<br /><br /> if(status=~"[12][0-9][0-9]")<br /> {<br /><br /><br /> if(ds_is_from_list("1"))<br /> {<br /> xlog("L_INFO", "Trying to change CONTACT FIELD\n");<br /> remove_hf("Contact");<br /> $avp(contact)="sip:"+$tU+"@PRO.XY.PUB.IP:5060";<br /> insert_hf("Contact:<$avp(contact)>\r\n","Call-ID");<br /><br /> }<br /> }<br />}<br /><br />But it has no shanges with Contact (so strange, but I've read - it bug)Anonymoushttps://www.blogger.com/profile/05348398080743545994noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-76002284166929723252014-04-25T02:43:29.463-07:002014-04-25T02:43:29.463-07:00Try to remove nat_uac_test("19") from ro...Try to remove nat_uac_test("19") from route[NATMANAGE] so You will allways be treated like behind NAT, You will see in kamctl ul show -> Cflag:: 64.Anonymoushttps://www.blogger.com/profile/17693019071971300123noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-51405442243440987542014-04-25T02:41:04.860-07:002014-04-25T02:41:04.860-07:00There should be more then one usage od rtpproxy_ma...There should be more then one usage od rtpproxy_manage() in Your route, so when You use It second time it add IP to c=IN of SDP. Only place where rtpproxy_manage() should be is route[RTPPROXY].Anonymoushttps://www.blogger.com/profile/17693019071971300123noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-52278490932247629172014-02-27T00:24:16.210-08:002014-02-27T00:24:16.210-08:00Dear Mickael, In my example users are to be manage...Dear Mickael, In my example users are to be managed in Kamailio DB,they'll get registered at Kamailio and Asterisk give services only for Media level stuff. <br /><br />For setting up with A2billing you'll need to change this approach and see the Kamailio Asipto blog and integrate the A2billing SIP user table with kamailio directly and let the rest of the DB be used by asterisk(s). That way from A2billing Web-Panel you'll be able to manage SIP users/routes/LCR etc and Kamailio will be the only visible component of your project for each SIP endpoint.Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-6996429372952703982014-02-19T01:52:03.074-08:002014-02-19T01:52:03.074-08:00Hello Ahmed,
In your setup, how do I manage users...Hello Ahmed,<br /><br />In your setup, how do I manage users login/password? by kamailio or Asterisk?<br />I see only one peer in your Asterisk configuration.<br />I'm a little lost. :-)<br /><br />How to set up this configuration with a Ast2Billing system or users are managed by 1 SIP user (A, B, etc) = 1 SIP peer/friend in asterisk.<br /><br />Thank you,<br />MickaelMickaelhttps://www.blogger.com/profile/02469309983869235886noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-33957785484788596962014-01-31T13:57:16.730-08:002014-01-31T13:57:16.730-08:00Hi Gohar,
Thanks for a detailed and informative d...Hi Gohar,<br /><br />Thanks for a detailed and informative descriptions of the setup. I'm also trying to setup similar network, without the LB for this time.<br /><br />I'm running into an issue, I'm not sure whether you've seen this yourselves. Please share some pointers. My network is:<br /><br />clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch)<br /><br />The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But. it's not. As a result, ACK from Caller is not reaching back to Kamailio.<br /><br />How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg.<br /><br />SIP 200 OK -><br /><br />Via: SIP/2.0/UDP 10.1.128.11;branch=z9hG4bKa7ea.4013d9881c1b7fe7b4c6c0f0e8f9d6b6.0<br />Via: SIP/2.0/UDP :5060;rport=5060;branch=z9hG4bK-383736-9b4621118533d3ccea17992738433249<br />Record-Route: <br />Record-Route: ;r2=on;lr=on;ftag=791b5ae3;nat=yes><br />From: \"Dipak Biswas\" >;tag=791b5ae3<br />To: >;tag=atXF5443gQj9p<br />Call-ID: 20d5d6d366fcb06c259a0895b3e44b52@0:0:0:0:0:0:0:0<br />CSeq: 2 INVITE<br />Contact: <br />User-Agent: FreeSWITCH mod_sofia/1.4.2+git~20140108T200418Z~d8fc8469b4~64bit<br />Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY<br />Supported: timer, precondition, path, replaces<br />Allow-Events: talk, hold, conference, refer<br />Content-Type: application/sdp<br />Content-Disposition: session<br />Content-Length: 269<br /><br /><br />Thanks,<br />DipakAnonymousnoreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-8656397997766254302013-10-16T05:29:25.695-07:002013-10-16T05:29:25.695-07:00Hello Gohar...i tested this more than one time and...Hello Gohar...i tested this more than one time and i'm receiving the same error message in asterisk <br />...................................................................................<br />[2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)", ...): Name or service not known<br />[2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 192.168.1.80192.168.1.80'<br />[2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)...<br />..............................................................................<br />So help me plz to solve this issue...Anonymoushttps://www.blogger.com/profile/10264960809310116719noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-56827801812113366122013-10-10T02:22:00.231-07:002013-10-10T02:22:00.231-07:00Hello,
I'm trying to setup Kamailio and Aster...Hello,<br /><br />I'm trying to setup Kamailio and Asterisk on the samebox with Kamailio listening on a public + private IP and asterisk only on private IP. <br /><br />Unfortunately I'm unable to get working as expected. We need somebody get us an hand !<br />Please contact me at scramattegmailcom <br /><br />Best regardsUnknownhttps://www.blogger.com/profile/03362746953299986840noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-31990514410010500632013-10-04T12:15:48.536-07:002013-10-04T12:15:48.536-07:00As for the AVP error check ur avp configuation as ...As for the AVP error check ur avp configuation as follows....<br />modparam("dispatcher", "dst_avp", "$avp(i:271)")<br />modparam("dispatcher", "grp_avp", "$avp(i:272)")<br />modparam("dispatcher", "cnt_avp", "$avp(i:273)")Anonymoushttps://www.blogger.com/profile/10264960809310116719noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-10946272679783225132013-10-01T13:03:52.726-07:002013-10-01T13:03:52.726-07:00Hi Ahmed,
first- thank you for this excellent blog...Hi Ahmed,<br />first- thank you for this excellent blog. I am starting to learn about kamailio (I am asterisk admin) and I have one question. Is it "correct" to use kamailio with sip peers and asterisk which are all in private lan (no UA coming from wan side). Kamailio would have one interface point to provider, also with private IP address from provider, and one interface in subnet together with couple of asterisk servers and sip peers. In that way peers (phones) would register to kamailio, which would load balancing calls between asterisk servers and pass calls from asterisks to provider? Thx again! Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-8667122351574323122013-09-01T06:38:15.175-07:002013-09-01T06:38:15.175-07:00Like I said read the link and put the right value ...Like I said read the link and put the right value in the nat_test function, could be like (31) or so. I suggest take a full packet trace to know from which IP this GSM SIP traffic is coming, is it Public? or Private ?<br />Also post the situation on User mailing list to get a more accurate solution.Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-19914651882399971392013-08-31T11:08:48.960-07:002013-08-31T11:08:48.960-07:00no. 10.133.19.49 is GSM private IP ! 192.168.30.25...no. 10.133.19.49 is GSM private IP ! 192.168.30.250 is kamailio lan ipneiroman2khttps://www.blogger.com/profile/14086713009011736028noreply@blogger.comtag:blogger.com,1999:blog-7979378259821020425.post-36246568914761666452013-08-31T04:36:23.683-07:002013-08-31T04:36:23.683-07:00Read this:
http://www.kamailio.org/docs/modules/4...Read this: <br />http://www.kamailio.org/docs/modules/4.0.x/modules/nathelper.html#idp15376536<br />This is really tough then...hmmm...Your "Public IP" is again a Private IP. !!?<br /><br />Gohar Ahmedhttps://www.blogger.com/profile/00614288169644917715noreply@blogger.com