Mapping RTP to correct SIP Call-ID
Posted: 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 ?
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 ?