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

[Serhat Guler]

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 <albertollamaso at 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 <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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>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> 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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