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.

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 list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com