Missing First Part of RTP Stream?

I’m having some trouble with my Asterisk box. My version is 1.4.35. The server sits inside the network behind an RV042 router. I have only SIP trunks through Broadvox so no Analog lines. My phones are all Aastra 57is.

Some of my calls seem to be missing the first part of the audio stream, at least that is what seems. Not all of them, but about 10% of them. The users dial the number and hear just dead air until suddenly there is audio. For example, I may dial a number and it’s just dead air until I hear “…help you? Hello?” No ringing and I miss the first part of the conversation. Or I may hear nothing and then someone’s voicemail picks up. I’ve captured the packets and I see:

Phone <-----> Server
-----> Invite SDP
<----- 407 Proxy Auth Req.
-----> ACK
-----> Invite SDP
<----- Trying
<----- 183 Session Progress SDP
-----> RTP
<----- RTP
and then the rest of the stuff.

Why would I not hear the first part of the conversation? I haven’t listened to the media stream yet since the server’s at a remote site. And should I worry about Comfort Noise being generated? I keep getting the warning that it’s isn’t fully implemented.

Thanks in advance!

The call for which you provided the trace hasn’t been answered, so doesn’t match your description of the problem.

183 with SDP is a request to set up one way audio for in-band call progress tones and messages.

I’d like to provide you with the most information possible so how is the best way to post my wireshark info? Just type it in like I was doing or is there a better way? And do you need the whole thing or just up to a point? I can’t say I’ve ever uploaded wireshark info before.

TIME| Phone <--------------> Server <----------------> Broadvox
521.745| Invite SDP ---------->
521.745| _________<---------- 407 Proxy Auth Req.
521.859| ACK ----------------->
521.880| Invite SDP ---------->
521.880| _________<---------- 100 Trying
521.898| __________________Invite SDP ------------->
521.961| __________________________ <------------- 100 Trying
523.349| __________________________ <------------- 183 Session Progress SDP
523.349| _________ <---------- 183 Session Progress SDP
523.574| RTP ----------------->
523.574| _________________ RTP ---------------------->
523.831| __________________________ <-------------- RTP
523.831| _________ <----------- RTP
524.470| RTP (CN) ------------>
555.604| __________________________ <------------- 200 OK SDP
555.605| _________________ ACK --------------------->
555.605| _________ <---------- 200 OK SDP
555.790| ACK ------------------>
555.815| RTP ------------------>
555.815| __________________ RTP ------------------->
Then there’s a bunch of RTP and RTP(CN). All the RTP other than the RTP(CN) is RTP(g711U).

Is it because of the gap between 524.470 and 555.604? I’ve been working on Asterisk systems for about a year now, but I’m definately not an expert. Thanks for any help you can give!

EDIT: Edited for formatting. Didn’t like the spaces…

The 200 OK from Broadvox may be a retransmission. In any case, there is no report of any RTP from them after this in your trace. One cannot tell from the trace when that 200 OK should have been received.

Then what can I post to give you enough information to help?

Firstly, the most likely cause is that you have an overloaded network and it is simply dropping the odd packet, and it is happening to drop the first 200 OK from Broadvox on occasions.

You really need timestamped logging from the Broadvox end, so that you know when they first tried to indicate that the call was answered.

OK, so here’s what I’ve found.

Broadvox shows that they are sending the RTP stream and their capture shows ringing in it.
I attached a hub outside the router and captured and showed there is ringing in the rtp stream.
I attached the hub outside the asterisk box and the rtp stream doesn’t even start until there is voice, not ringing.

Since the stream appears correct outside the router but not inside the router, it seems the RV042 is cutting it somehow. I have SPI turned off and I don’t see any errors or entries in the logs. We use RV042 routers in most of our clients since they are stable and offer good VPN functionality. However, if they are not good for VOIP traffic, what is a good alternative? I’ve seen several posts across different forums that say the RV042 works good for VOIP. Anyone have suggestions?

Thanks for all the input so far!