18 Feb
2019
18 Feb
'19
10:07 a.m.
Hi Guys,
I have recently setup a Kamailio server which can accept SIP or WebSocket
connections but voice is not coming in the case of SIP client.
I have no idea to change avp in the incoming request, please advise how to
proceed.
18 Feb
18 Feb
2:42 p.m.
Hello,
On 18.02.19 10:07, Adesh Pandey wrote:
Hi Guys,
I have recently setup a Kamailio server which can accept SIP or
WebSocket connections but voice is not coming in the case of SIP client.
I have no idea to change avp in the incoming request, please advise
how to proceed.
the details you provide here doesn't give any clue of what is the real
problem, where and why, so likely nobody can really help directly at
this stage.
I suggest to start with debug=3 in kamailio cfg and watch the logs when
you do testing calls. Watch also the web browser console/diagnostic
tools to see if you get any hints there.
If you search on the web, you should fine some tutorials about using
kamailio in webrtc -- they might be a good reference to compare with
your config.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC,
USA -- www.asipto.com
20 Feb
20 Feb
8:51 a.m.
Hi
I have configured Kamailio as dispatcher for Asterisk, if I am logging in
to kamilio with webrtc can make calls with Asterisk and everything goes
fine but when I try to make calls from Zoiper my asterisk box says that :
[2019-02-19 08:45:15] WARNING[4649][C-00000008] chan_sip.c: Failed to
receive SDP offer/answer with required SRTP crypto attributes for audio
[2019-02-19 08:45:15] NOTICE[4649][C-00000008] chan_sip.c: Received AVP
profile in audio answer but AVPF is enabled, disabling: audio 8000 RTP/AVP
106 9 3 111 0 8 97 110 112 98 101 100 99 102
[2019-02-19 08:45:15] VERBOSE[4649][C-00000008] netsock2.c: Using SIP RTP
CoS mark 5
[2019-02-19 08:45:15] VERBOSE[4649][C-00000008] res_rtp_asterisk.c: DTLS
ECDH initialized (secp256r1), faster PFS enabled
I have explored over the internet but nothing found, please help me to
modify AVP in such a way that RTP request gets processed by Asterisk.
On Mon, Feb 18, 2019 at 7:12 PM Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
On 18.02.19 10:07, Adesh Pandey wrote:
Hi Guys,
I have recently setup a Kamailio server which can accept SIP or
WebSocket connections but voice is not coming in the case of SIP client.
I have no idea to change avp in the incoming request, please advise
how to proceed.
the details you provide here doesn't give any clue of what is the real
problem, where and why, so likely nobody can really help directly at
this stage.
I suggest to start with debug=3 in kamailio cfg and watch the logs when
you do testing calls. Watch also the web browser console/diagnostic
tools to see if you get any hints there.
If you search on the web, you should fine some tutorials about using
kamailio in webrtc -- they might be a good reference to compare with
your config.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in
Washington, DC, USA -- www.asipto.com
10:36 a.m.
Could you share your Kamailio config.
I now troubleshoot call to WebRTC clients, and this will help me.
ср, 20 февр. 2019 г., 10:52 Adesh Pandey <adesh.pandey(a)myoperator.co>co>:
Hi
I have configured Kamailio as dispatcher for Asterisk, if I am logging in
to kamilio with webrtc can make calls with Asterisk and everything goes
fine but when I try to make calls from Zoiper my asterisk box says that :
[2019-02-19 08:45:15] WARNING[4649][C-00000008] chan_sip.c: Failed to
receive SDP offer/answer with required SRTP crypto attributes for audio
[2019-02-19 08:45:15] NOTICE[4649][C-00000008] chan_sip.c: Received AVP
profile in audio answer but AVPF is enabled, disabling: audio 8000 RTP/AVP
106 9 3 111 0 8 97 110 112 98 101 100 99 102
[2019-02-19 08:45:15] VERBOSE[4649][C-00000008] netsock2.c: Using SIP RTP
CoS mark 5
[2019-02-19 08:45:15] VERBOSE[4649][C-00000008] res_rtp_asterisk.c: DTLS
ECDH initialized (secp256r1), faster PFS enabled
I have explored over the internet but nothing found, please help me to
modify AVP in such a way that RTP request gets processed by Asterisk.
On Mon, Feb 18, 2019 at 7:12 PM Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
On 18.02.19 10:07, Adesh Pandey wrote:
Kamailio (SER) - Users Mailing
List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Guys,
I have recently setup a Kamailio server which can accept SIP or
WebSocket connections but voice is not coming in the case of SIP
client.
I have no idea to change avp in the incoming request, please advise
how to proceed.
the details you provide here doesn't give any clue of what is the real
problem, where and why, so likely nobody can really help directly at
this stage.
I suggest to start with debug=3 in kamailio cfg and watch the logs when
you do testing calls. Watch also the web browser console/diagnostic
tools to see if you get any hints there.
If you search on the web, you should fine some tutorials about using
kamailio in webrtc -- they might be a good reference to compare with
your config.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in
Washington, DC, USA -- www.asipto.com
_______________________________________________
1901
days inactive
1903
days old
3 comments
3 participants
participants (3)
-
Adesh Pandey -
Daniel-Constantin Mierla -
Sergey Safarov