I am working on a Browser based SIP client. I have a SIP server
PortaOne which doesn't support Web Sockets. So I have decided to use
SIP-Proxy.
I have installed OverSIP on my linux ubuntu 14.04 server. and for WebRTC SIP client I am using JSSIP.
I register SIP accounts successfully but when I call from my web
browser it successfully dial the call but it disconnect the call when
call picked up on destination.
Please guide me if there is any configuration required in oversip to resolve this issue.
Thanks
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Does PortaOne support WebRTC media? I don't mean WebSocket but WebRTC media (ICE, DTLS, SRTP).Please, confirm.
El 31 ene. 2018 11:30, "Yasir Ikram" <mail....@gmail.com> escribió:
Hi,--I am working on a Browser based SIP client. I have a SIP server PortaOne which doesn't support Web Sockets. So I have decided to use SIP-Proxy.
I have installed OverSIP on my linux ubuntu 14.04 server. and for WebRTC SIP client I am using JSSIP.
I register SIP accounts successfully but when I call from my web browser it successfully dial the call but it disconnect the call when call picked up on destination.
Please guide me if there is any configuration required in oversip to resolve this issue.
Thanks
You received this message because you are subscribed to the Google Groups "OverSIP" group.
To unsubscribe from this group and stop receiving emails from it, send an email to oversip+u...@googlegroups.com.
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <SIP Request 9001360> applying outgoing Outbound support
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <UserAssertion module> user asserted, adding P-Asserted-Identity for SIP Request 9001360
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <SIP Request 9001360> replying 100 "Trying"
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <WsFraming> sending text frame: payload_length=267
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <Proxy proxy_out 9001360> destination found in the DNS cache
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <Proxy proxy_out 9001360> trying single target: udp:78.40.244.5:5060
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <Proxy proxy_out 9001360> received response 180
Jan 30 06:51:59 s112209 oversip[26917]: INFO: <SipEvents> [user] on_provisional_response: 180 'Ringing'
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <SIP Request 9001360> forwarding response 180 "Ringing"
Jan 30 06:51:59 s112209 oversip[26917]: DEBUG: <WsFraming> sending text frame: payload_length=616
Jan 30 06:52:03 s112209 oversip[26917]: DEBUG: <NICT 8cc66e4f9f2752160186063b0a316ba3da09c31e> timer K expires, transaction terminated
Jan 30 06:52:04 s112209 oversip[26917]: DEBUG: <NICT 754089bd9f2752160186063b0a316ba3da09c31e> timer K expires, transaction terminated
Jan 30 06:52:14 s112209 oversip[26917]: DEBUG: <ICT 69e8fce34b152552f7bd81de1af30d89da6ca49e> sending ACK for [3456]XX response
Jan 30 06:52:14 s112209 oversip[26917]: DEBUG: <Proxy proxy_out 9001360> received response 500
Jan 30 06:52:14 s112209 oversip[26917]: INFO: <SipEvents> [user] on_failure_response: 500 'Expired'
Jan 30 06:52:14 s112209 oversip[26917]: DEBUG: <SIP Request 9001360> forwarding response 500 "Expired"
Jan 30 06:52:14 s112209 oversip[26917]: DEBUG: <WsFraming> sending text frame: payload_length=585
Jan 30 06:52:15 s112209 oversip[26917]: DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=411
Jan 30 06:52:15 s112209 oversip[26917]: DEBUG: <WsSipApp> received WS message: type=text, length=411
Jan 30 06:52:15 s112209 oversip[26917]: DEBUG: <IST 9001360> ACK received during completed state, now confirmed
Jan 30 06:52:16 s112209 oversip[26917]: INFO: <SipEvents> [user] INVITE from sip:20...@107.150.61.226 (UA: sipcli/v1.8) to sip:4309441923
Hi Yasir,You are not giving any information at all. If the call disconnects, have you checked the logs in the SIP client or in the SIP server, or even in OverSIP?Help yourself by checking the logs in the endpoints first (SIP client & Server). I doubt there is an issue with the proxy (OverSIP) here but it's impossible to know since no data is provided.
2018-01-31 11:30 GMT+01:00 Yasir Ikram <mail....@gmail.com>:
Hi,I am working on a Browser based SIP client. I have a SIP server PortaOne which doesn't support Web Sockets. So I have decided to use SIP-Proxy.
I have installed OverSIP on my linux ubuntu 14.04 server. and for WebRTC SIP client I am using JSSIP.
I register SIP accounts successfully but when I call from my web browser it successfully dial the call but it disconnect the call when call picked up on destination.
Please guide me if there is any configuration required in oversip to resolve this issue.
Thanks
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--José Luis Millán