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)
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