1 Nov
2016
1 Nov
'16
3:39 p.m.
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(a)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(a)gmail.com>
wrote:
--
Alberto Llamas
Phone: +1-786-805-6003
Telecommunications Engineer
Digium Certified Asterisk Professional (dCap)
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
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--
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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