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.
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
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@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
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@myoperator.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@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
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