Hello,
On 12/1/09 4:01 AM, tony at everestbn.com wrote:
> Hello,
>
> I'm having random audio issues (<10% of the
> time) with blind call
> transfers on a Polycom phone. According
> to RTPPROXY's logs, it seems it is substituting the same IP and port for
> the caller's and callee's address causing one end to hear their own voice
> while the other end hears nothing.
>
Is this real scenario? I mean, caller's and callee's IP and port for
media are the same? i haven't encounter such case to see if rtpproxy can
deal with... The idea is that rtpproxy learns the remote IP and port for
each side based on first rtp packet sent to local rtpproxy allocated ports.
Cheers,
Daniel
> INFO:rxmit_packets: caller's
> address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP
> port for caller to be 17857
> INFO:rxmit_packets: callee's address
> filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port
> for callee to be 17857
>
> The SDP in the SIP messages has all the
> correct IPs and ports and it appears from a SIP overview everything should
> work fine.
>
> Here is some background information:
> I am using
> Kamailio 1.5.3 with RTPPROXY 1.2.1.
>
> SIP PATH is as follows:
> PSTN Gateway---Kamailio--SIP Proxy Server--Polycom Phone
>
> RTP
> PATH:
> PSTN Gateway---Kamailio----Polycom Phone
>
> *There are
> nor firewalls or filters.
>
> Call Flow:
> I place a call from
> the gateway to the ip phone. Call is answered. When I do a
> blind transfer from the Polycom, the SIP Proxy sends an invite with no
> SDP, the gateway responds with a 200 with SDP and the SIP Proxy then sends
> an ACK with SDP. As I mentioned, this works most of the time and the
> SIP messages look identical for when the transfer works and when it
> doesn't. When it doesn't work the gateway end hears their own voice while
> the recipient Polycom transfer (usually another phone on the same proxy)
> hears nothing.
>
> Here is a portion of my configuration file:
> .....
> if (is_method("INVITE")) {
>
> if((search("^Content-Type:[ ]*application/sdp")) ||
> (search("^Content-Type:application/sdp"))){
> rtpproxy_offer("fcr");
>
> setflag(12);
> }
> }
>
> if
> (is_method("ACK")) {
>
>
> if((search("^Content-Type:[ ]*application/sdp")) ||
> (search("^Content-Type:application/sdp"))) {
>
> rtpproxy_answer("fcr");
>
> }
> }
>
>
> t_on_reply("1");
> ....
> }
>
> onreply_route[1]
> {
> if (status=~"(180)|(183)|(2[0-9][0-9])"){
> if((search("^Content-Type:[ ]*application/sdp"))
> ||
> (search("^Content-Type:application/sdp"))) {
> if (isflagset(12)) {
>
> rtpproxy_answer("fcr");
> } else {
> rtpproxy_offer("fcrl");
> }
> }
> }
> }
>
> Any help
> would be greatly appreciated!
>
> Thanks,
>
> Tony
>
>
>
>
>
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>
--
Daniel-Constantin Mierla
* http://www.asipto.com/