Hi Daniel, Here is something i traced in the log:
ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force' ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000] What's the cause of this error? i am using code from the master branch. Perhaps this has something to do with the rptengine service crash/termination. Regards On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini < abhishek.sa...@enukesoftware.com> wrote: > Hi Daniel, > > As you instructed, i installed kamailio from the master branch (which has > rtpengine module). Along with this, i installed the rtpengine package from > sipwise, as instructed by them. > > I also updated this param : modparam("nathelper", "sipping_from", " > sip:pin...@abc.com") to my domain > > Now the scenario is as follows: > > 1) I am able to call webrtc(firefox and chrome) from iphone, the > signalling seems to be working fine, call can be paused, resumed etc.., but > there is no audio/video transmission. > > 2) Still when i call from webrtc to iphone - the retpengine service of > ubuntu terminates/crashes (like before) and needs to be restarted. > > Does it have any thing to do with rtp port ranges? or is there some other > misconfiguration? > > > Regards, > Abhishek > > > > > On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla < > mico...@gmail.com> wrote: > >> Hello, >> >> maybe you should play with kamailio master branch (which is in testing >> phase before becoming 4.2) -- there you have the rtpengine -- and see if >> you get it working. Once that, you can look at using an older version, >> knowing you have it working and be able to compare. As I needed latest >> features, whenever I needed webrtc gatewaying, I used devel branch of >> rtpengine module. >> >> Cheers, >> Daniel >> >> >> On 16/09/14 14:24, Abhishek Saini wrote: >> >> Hi Daniel, >> >> >> I was able to solve a fraction of my problem, Actually, the github link >> had used rtpengine.so and i was using rptproxy-ng.so, there is a difference >> in the flag conventions between the two; i modified that to achieve a >> little progress. >> >> Now, i am able to call on webrtc(firefox) from sip phone. However, after >> accepting call, there is no audio, and disconnecting the call from either >> end does not disconnect the call. >> >> When i try to call from webrtc(firefox) to sip phone, there is no >> signalling at all, and the sip phone to webrtc calls can't connect after >> that. (I analyzed that mediaproxy-ng/rtpengine process terminates and has >> to be started again) >> >> Following are the links to my latest kamailio.cfg file and port trace >> log of sip messages. >> http://jmp.sh/o0apKgP >> http://jmp.sh/HXnFRQj >> >> I am clueless at the moment! >> >> Regards, >> Abhishek >> >> >> >> On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini < >> abhishek.sa...@enukesoftware.com> wrote: >> >>> Hi Daniel, >>> >>> Thanks for this. >>> >>> I took the entire config files and configured it as per my ips and >>> ports, after doing that, still no call establishment(webrtc to classic sip >>> phones and vice-versa). Following is what i get in kamailio.log: >>> >>> rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it >>> enabled >>> ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown >>> option ` ' >>> ERROR: <script>: ==> duri=[ >>> sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp] >>> INFO: <script>: Request coming from WS >>> ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call(): unknown >>> option ` ' >>> INFO: <script>: Reply from softphone: 100 >>> >>> And this SIP message: >>> SIP/2.0 603 Failed to get local SDP. >>> >>> Regards, >>> Abhishek >>> >>> >>> >>> >>> On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla < >>> mico...@gmail.com> wrote: >>> >>>> Hello, >>>> >>>> the reply code indicates that the media type is not supported, thus >>>> there has been no gatewaying between webrtc and classic rtp. Just replacing >>>> rtpproxy with rtpengine is not enough, there are different parameters that >>>> have to be provided. >>>> >>>> Searching on web, I see that Carlos has published a config for it, see: >>>> - https://github.com/caruizdiaz/kamailio-ws >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 15/09/14 12:58, Abhishek Saini wrote: >>>> >>>> Hi, >>>> >>>> I have successfully setup rtpproxy-ng kamailio module and >>>> mediaproxy-ng package on my ubuntu box. As suggested here: >>>> http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html >>>> >>>> I have kept rtpproxy-ng's configuration same as the rtpproxy module, >>>> but still not able to connect the webrtc calls to classic sip phones (and >>>> vice-versa). Below is the sip message that is traced: >>>> >>>> >>>> SIP/2.0 488 Not acceptable here. >>>> Via: SIP/2.0/TCP >>>> 54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$ >>>> Via: SIP/2.0/WS >>>> df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$ >>>> From: "admin" <sip:ad...@abc.com>;tag=bzhwwG8nT2gFwwJgIyrz. >>>> To: <sip:h...@abc.com>;tag=OIllTQf. >>>> Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a. >>>> CSeq: 65463 INVITE. >>>> User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2). >>>> Supported: replaces, outbound. >>>> Content-Length: 0. >>>> >>>> Can you please let me know, what's going wrong and how can i proceed. >>>> >>>> Regards, >>>> Abhishek >>>> >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - >>>> http://www.linkedin.com/in/miconda >>>> Next Kamailio Advanced Trainings 2014 - http://www.asipto.com >>>> Sep 22-25, Berlin, Germany >>>> >>>> >>> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - >> http://www.linkedin.com/in/miconda >> Next Kamailio Advanced Trainings 2014 - http://www.asipto.com >> Sep 22-25, Berlin, Germany >> >> >
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