![]() ![]() On the sip call flow graph, we can check RTP direction and codec.Use 'rtp' as the expression to filter RTP packets.Is the RTP stream be sent with right ptime?.Is the RTP stream be decoded in the right codec?.Is the RTP stream send and receive on the right IP address and port?.When we have a voice issue, we could check the following problem with Wireshak: Contact: the address for the subsequent request.There are two parts in the sip INVITE request, SIP headers, and SDP. See the following figure about the SIP call filtered by Call-ID.Įnable display raw for SIP message so that we don't need to expand every sip header or SDP parameters. Usually, SIP entity will generate the random call-id string for each call, so we can mark one sip call with the call-id parameter. In SIP protocol, we can use call-id, from-tag, to-tag to identify a call. Direction, source and dest port of RTP stream.SIP signaling flow between different UA.Click the Flow Sequence button we can see the graph of this call with some details: Select the calls you want to check, then we can see the invalid option Flow Sequence become available. ![]()
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