![]() We can see the RTP player after click the Play Streams button.Who needs the Wireshark GUI right let’s do this at the command line and be grown up about things. Play one RTP stream, in the RTP Streams list, Analyze > Play Streams.Select and Play Stream in the call list.we can decode the UDP packets to RTP manually.įor now, Wireshark only supports playing pcmu and pcma codec. The media line of the SDP message in INVITE or 200OK sip messagesĪs we know RTP usually uses UDP transport, when the sip call flow in the PCAP file is incomplete the Wireshark may not parse the UDP packets to RTP streams.On the SIP call flow graph, we can see the source and dest port of one RTP stream.We can see all the RTP streams display and we can see some information of these RTP streams, like source port and dest port, SSRC, payload, max delta, lost percentage of the packets and jitter.īut how could we know which stream is the one we want to check? Use the menu 'Telephony > RTP > RTP Streams'.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.
0 Comments
Leave a Reply. |
AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |