If you need a real answer, then you need to show the message exchange for the calls that did not work I guess. What probably is wrong here is that "inline" is in a new line but I am sure it's just your mailer wrapping the lines. I would first look at the crypto-suites if they are properly supported by the phones. They are AES_CM_128_HMAC_SHA1_80, AES_CM_128_HMAC_SHA1_32, F8_128_HMAC_SHA1_32 currently. If the Polycom can't support AES_CM_128_HMAC_SHA1_80, it could always choose to go unencrypted unless you configured it to require SRTP.
On 06/24/2011 10:07 PM, Kumaran wrote: > Hi All, > Please check my scenario > 1.user 200-Polycom SRTP enabled > 2.user 420-X-lite > 3.200 calls 420 > 4.420 answers the call > 5.Call will be in SRTP > 6.X-lite phones sends SRTP in SDP part > > v=0 > o=- 7 2 IN IP4 176.25.3.197 > s=CounterPath X-Lite 3.0 > c=IN IP4 176.25.3.197 > t=0 0 > m=audio 1884 RTP/SAVP 0 8 101 > a=crypto:2 AES_CM_128_HMAC_SHA1_80 > inline:XxIabqd8zQIHcS89lsCDBPD0S1INE0ZNxm9Zsqxd > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > m=audio 2494 RTP/AVP 0 8 101 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > > Whether this is correct?If its works then why its not working for other > polycom? > > Regards, > Kumaran T > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
