Thanks for the recommendations Alberto. I'll definitely try it out and
hopefully will be able to call a softphone from webrtc client.
Cheers,
Serhat
On 1 November 2016 at 15:17, Alberto Llamas <albertollam...@gmail.com>
wrote:
> Hi Serhat,
>
> If you take a look of SDP body of your INVITES you will note that you are
> offering SRTP.
>
> What you should do from my point of view is detect when an INVITE from the
> sipml5 softphone goes to the ims-softphone or other end-point which you are
> aware doesn't support SRTP and use the RTPEngine module (combination
> of rtpengine_offer and rtpengine_answer functions).
>
> *When INVITE (sipml5 -> Kamailio -> endPoint with out SRTP) you should
> perform something like this:*
>
> rtpengine_offer("trust-address replace-origin replace-session-connection
> ICE=remove RTP/AVP");
> t_on_reply("3");
>
> *Then for replies you will need something like the route:*
>
> onreply_route[3] {
>
> if (t_check_status("183")) {
> change_reply_status("180", "Ringing");
> remove_body();
> exit;
> }
>
> if(!(status=~"[12][0-9][0-9]") || !(sdp_content()))
> return;
> rtpengine_answer("trust-address replace-origin
> replace-session-connection ICE=force");
>
> route(NATMANAGE);
> }
>
> Or other approcah if offer SRTP and when the other ends answer with a
> 488 Not Supported Here do the same.
>
> It is one way of bridging SRTP->RTP with the RTPEngine module.
>
> Regards,
>
>
> On Tue, Nov 1, 2016 at 2:48 PM, Serhat Guler <srtgu...@gmail.com> wrote:
>
>> 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/AAAOOv4h2ci7bJu
>>>> uJvSbs3poa?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
>>>
>>>
>>
>> _______________________________________________
>> 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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