1 Nov
2016
1 Nov
'16
2:13 p.m.
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.
1 Nov
1 Nov
2:15 p.m.
New subject: Fwd: Kamailio + IMS + sipml5 call to softphone
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
2:38 p.m.
New subject: Fwd: Kamailio + IMS + sipml5 call to softphone
HI Serhat,
Is it possible to have a packet capture for the cases you mention.
Regards,
On Tue, Nov 1, 2016 at 12:15 PM, Serhat Guler <srtguler(a)gmail.com> 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 list
sr-users(a)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)
2:39 p.m.
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
<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(a)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
2:45 p.m.
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:
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@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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4:48 p.m.
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(a)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(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)
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
5:17 p.m.
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 <srtguler(a)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 <albertollamaso(a)gmail.com>
wrote:
--
Alberto Llamas
Phone: +1-786-805-6003
Telecommunications Engineer
Digium Certified Asterisk Professional (dCap)
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:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 <miconda(a)gmail.com
--
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
9:27 p.m.
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 <albertollamaso(a)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 <srtguler(a)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 <albertollamaso(a)gmail.com>
wrote:
--
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 <
miconda(a)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
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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
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)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
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--
Alberto Llamas
Phone: +1-786-805-6003
Telecommunications Engineer
Digium Certified Asterisk Professional (dCap)
_______________________________________________
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8 comments
3 participants
participants (3)
-
Alberto Llamas -
Daniel-Constantin Mierla -
Serhat Guler