[SR-Users] Kamailio + IMS + sipml5 call to softphone

[Alberto Llamas]

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 <srtguler at 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 <miconda at 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-users at 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
>>
>>
>> _______________________________________________
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>> sr-users at lists.sip-router.org
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>>
>>
>
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>


-- 
Alberto Llamas
Phone: +1-786-805-6003
Telecommunications Engineer
Digium Certified Asterisk Professional (dCap)
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