How do I debug SIP?
To debug SIP messages, use the debug ccsip command. This command has several options, as Example 4-13 shows. Use messages to see the SIP method and response messages, as shown previously in Example 4-1. The media option shows RTP information.
How do I check my Cisco SIP status?
SIP Trunk status can be monitored by configuring an out-of-dialog (OOD) SIP Options PING as a keepalive mechanism on the dial-peer(s) pointing towards the SIP Trunk, using the CLI example below. When calls to the SIP trunk are successful, the dial-peer is in “active” state.
Does Cisco use SIP?
Cisco has a SIP proxy server product. Redirect server—UAs and proxy servers can contact a redirect server to find the location of an endpoint.
What is SIP UA Cisco?
You can enable Session Initiation Protocol (SIP) UA commands, using the ios-voice:sip-ua configuration mode. The ios-voice:sip-ua mode is a part of Cisco-IOS-XE-voice module. The following operations are allowed in the ios-voice:sip-ua mode: Operations. X-path.
How does Cisco SIP work?
Cisco CallManager can act as both a server or client (a back-to-back user agent). SIP uses a request/response method to establish communications between various components in the network and to ultimately establish a call or session between two or more endpoints.
What is SIP server?
A SIP server, also known as a SIP Proxy, deals with all the management of SIP calls in a network and is responsible for taking requests from the user agents in order to place and terminate calls. SIP Servers are usually included inside of SIP-enabled IP-PBXs.
How a SIP call is setup?
Basic SIP session setup involves a SIP UA client sending a request to the SIP URL of the called endpoint (UAS), inviting it to a session. If the UAC knows the IP address of the UAS, it can send the request. Otherwise, the UAC sends the request to a proxy or redirect server to locate the user.
What do I need to know about the SIP protocol?
Basic knowledge about the SIP Protocol and the call flow Messages. 1. Call Flow between PBX to Cisco SIP IP Phone—Successful Setup and Disconnect Below diagram illustrates a successful gateway-to-Cisco SIP IP phone call setup and disconnect.
How do I identify a call session in Cisco SIP?
Cisco SIP IP phone A is identified as the call session initiator in the From field. A unique numeric identifier is assigned to the call and is inserted in the Call-ID field. The transaction number within a single call leg is identified in the CSeq field.
How do I know if my SIP IP phone supports media capability?
If Cisco SIP IP phone B supports the media capability advertised in the INVITE message sent by Cisco SIP IP phone A, it advertises the intersection of its own and Cisco SIP IP phone A’s media capability in the 200 OK response.
What happens if Cisco SIP IP phone b does not support media?
If Cisco SIP IP phone B does not support the media capability advertised by Cisco SIP IP phone A, it sends back a 400 Bad Request response with a 304 Warning header field. 4. Cisco SIP IP phone A sends a SIP ACK to Cisco SIP IP phone B. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone B.