Mapping RTP to correct SIP Call-ID

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mohitshri
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Joined: Wed Feb 11, 2015 5:09 am

Mapping RTP to correct SIP Call-ID

Post by mohitshri » Wed Feb 11, 2015 5:14 am

First of all, kudos to all present. Thanks to this site, I was able to strengthen my basics in SIP + RTP.
I have a query for which my approaches so far seem irrelevant.

Using Linphone, I am able to initiate and finish a voice call across two hosts. As a result, the following has already happened -
1. SIP "Call-ID" field was generated, thus I am able to recreate session flow.
2. RTP packets were transmitted, thus, 2 SSRCs (one each from src-->dst and vice versa) were obtained on Wireshark.

My question is - What if two hosts using two different SIP accounts (but merely 2 instances of the same user agent) are activated and the call done as before ? How do we find which RTP packet corresponds to which session initiated by the host ?

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Aaron
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Joined: Thu Jul 12, 2007 12:13 am

Re: Mapping RTP to correct SIP Call-ID

Post by Aaron » Wed Feb 11, 2015 8:52 am

The only reliable way you can match the RTP to the SIP dialogue is with the socket (IP address and port) specified in the SIP INVITE request and response.

If you do a packet capture with WireShark (www.iwreshark.org) it has an inbuilt tool to do this for you under the Telephony->VoIP Calls menu. To do it manually you need to capture the SDP payload in the INVITE packets and match it with the ports in use on the RTP packets.

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