Hi Alberto, Thanks for looking into this. In the expert settings of sipml5 it says that disabling RTCWeb Breaker should make it compatible with softphones which are not implementing SRTP, that's how I have been testing it though. May I ask what attribute you looked at to get to the conclusion you have ? Thanks.
Serhat On 1 November 2016 at 13:39, Alberto Llamas <albertollam...@gmail.com> wrote: > Hello Serhat, > > When you are using the webphone (sipml5) by WebRTC the media is secured > with SRTP. So if the other end-point supports SRTP usually you don't have > major issues. It is like when you communicate between two sipml5 web phones > A and B. > > But when you are trying to communicate to the IMS softphone, be sure that > the softphone supports SRTP otherwise you will need to configure a RTP > Proxy like RTPEngine in kamailio module in order to "translate" between > plain RTP and SRTP. > > This is what I see is your issue based on the pcap files. > > PS: You can have a setup in your kamailio config file to offer first SRTP > and if the other end-point doesn't support it (when you receive a 4XX > reply) then send a Re-INVITE with plain RTP. > > Regards, > > On Tue, Nov 1, 2016 at 12:45 PM, Serhat Guler <srtgu...@gmail.com> wrote: > >> Hi Daniel, hi Alberto, >> >> Thanks for your prompt replies. I have put 2 pcap files in dropbox ( >> https://www.dropbox.com/sh/fzclmbpniebrvx1/AAAOOv4h2ci7bJuuJvSbs3poa?dl=0 ) >> . trace.mercuro.pcap is the one where the session is set up, but there is >> no audio flow and trace.boghe.pcap is the one with 488 error. >> >> Cheers, >> Serhat >> >> On 1 November 2016 at 12:39, Daniel-Constantin Mierla <mico...@gmail.com> >> wrote: >> >>> Hello, >>> >>> can you get the SIP INVITE content that was received by the endpoint >>> returning 488? Maybe we can spot if there is something wrong in the sip >>> message content or an issue in the endpoint software. Maybe it doesn't like >>> headers with random string instead of ip addresses (e.g., in via, contact >>> ...). >>> >>> I am not aware of any ims softphone with webrtc capabilities. >>> Cheers, >>> Daniel >>> >>> >>> On 01/11/16 12:15, Serhat Guler wrote: >>> >>> Hi, >>> >>> I have a setup as follows: >>> >>> IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for >>> webrtc calls. >>> >>> Calls(both audio and video) between to sipml5 clients using firefox web >>> browser is possible. The session is setup for the calls from sipml5 to >>> Mercuro, but then there isn't audio flow as the codecs are not compatible. >>> >>> Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and >>> OPUS codecs as firefox but this time the session isn't being setup. Boghe >>> replies with "Reason: SIP; cause=488; text="Bad content" >>> " I have seen a similar issue has been mentioned here: >>> https://github.com/c00lz3r0/boghe/issues/157 but the initial invite >>> request from sipml5 does have the SDP with media attributes. >>> >>> >>> Any advice or are there any other IMS softphones that I can use to test >>> for this scenario. Thanks a lot. >>> >>> P.S. The previous email went out directly unintentionally. >>> Serhat >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - >>> http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > -- > Alberto Llamas > Phone: +1-786-805-6003 > Telecommunications Engineer > Digium Certified Asterisk Professional (dCap) > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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